* [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-29 13:33 ` Markus Armbruster
2019-01-29 15:52 ` Markus Armbruster
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 02/14] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
` (13 subsequent siblings)
14 siblings, 2 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Eric Blake, Markus Armbruster
This patch adds structures into qapi to replace the existing
configuration structures used by audio backends currently. This qapi
will be the base of the -audiodev command line parameter (that replaces
the old environment variables based config).
This is not a 1:1 translation of the old options, I've tried to make
them much more consistent (e.g. almost every backend had an option to
specify buffer size, but the name was different for every backend, and
some backends required usecs, while some other required frames, samples
or bytes). Also tried to reduce the number of abbreviations used by the
config keys.
Some of the more important changes:
* use `in` and `out` instead of `ADC` and `DAC`, as the former is more
user friendly imho
* moved buffer settings into the global setting area (so it's the same
for all backends that support it. Backends that can't change buffer
size will simply ignore them). Also using usecs, as it's probably more
user friendly than samples or bytes.
* try-poll is now an alsa backend specific option (as all other backends
currently ignore it)
AudiodevPaPerDirectionOptions struct might look pointless right now, but
later we will have new per-direction options, so it's more future-proof
this way.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Notes:
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
Makefile.objs | 6 +-
qapi/audio.json | 233 ++++++++++++++++++++++++++++++++++++++++++
qapi/qapi-schema.json | 1 +
3 files changed, 237 insertions(+), 3 deletions(-)
create mode 100644 qapi/audio.json
diff --git a/Makefile.objs b/Makefile.objs
index 67a054b08a..d9ee3fed32 100644
--- a/Makefile.objs
+++ b/Makefile.objs
@@ -1,6 +1,6 @@
-QAPI_MODULES = block-core block char common crypto introspect job migration
-QAPI_MODULES += misc net rdma rocker run-state sockets tpm trace transaction
-QAPI_MODULES += ui
+QAPI_MODULES = audio block-core block char common crypto introspect job
+QAPI_MODULES += migration misc net rdma rocker run-state sockets tpm trace
+QAPI_MODULES += transaction ui
#######################################################################
# Common libraries for tools and emulators
diff --git a/qapi/audio.json b/qapi/audio.json
new file mode 100644
index 0000000000..bd6e2494bd
--- /dev/null
+++ b/qapi/audio.json
@@ -0,0 +1,233 @@
+# -*- 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.
+
+##
+# @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')
+#
+# @try-poll: attempt to use poll mode, falling back to non polling access on
+# failure (default on)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevAlsaPerDirectionOptions',
+ 'data': {
+ '*dev': 'str',
+ '*try-poll': 'bool' } }
+
+##
+# @AudiodevAlsaOptions:
+#
+# Options of the alsa audio backend.
+#
+# @alsa-in: options of the capture stream
+#
+# @alsa-out: options of the playback stream
+#
+# @threshold: set the threshold (in microseconds) when playback starts
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevAlsaOptions',
+ 'data': {
+ '*alsa-in': 'AudiodevAlsaPerDirectionOptions',
+ '*alsa-out': 'AudiodevAlsaPerDirectionOptions',
+ '*threshold': 'uint32' } }
+
+##
+# @AudiodevDsoundOptions:
+#
+# Options of the dsound audio backend.
+#
+# @latency: add extra latency to playback in microseconds (default 10000)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevDsoundOptions',
+ 'data': {
+ '*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')
+#
+# @try-poll: attempt to use poll mode, falling back to non polling access on
+# failure (default on)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevOssPerDirectionOptions',
+ 'data': {
+ '*dev': 'str',
+ '*try-poll': 'bool' } }
+
+##
+# @AudiodevOssOptions:
+#
+# Options of the oss audio backend.
+#
+# @oss-in: options of the capture stream
+#
+# @oss-out: options of the playback stream
+#
+# @try-mmap: try using memory mapped access, falling back to non
+# memory mapped access on failure (default off)
+#
+# @exclusive: open device in exclusive mode (vmix won't work) (default off)
+#
+# @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': {
+ '*oss-in': 'AudiodevOssPerDirectionOptions',
+ '*oss-out': 'AudiodevOssPerDirectionOptions',
+ '*try-mmap': 'bool',
+ '*exclusive': 'bool',
+ '*dsp-policy': 'uint32' } }
+
+##
+# @AudiodevPaPerDirectionOptions:
+#
+# Options of the pa backend that are used for both playback and recording.
+#
+# @name: name of the sink/source to use
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevPaPerDirectionOptions',
+ 'data': {
+ '*name': 'str' } }
+
+##
+# @AudiodevPaOptions:
+#
+# Options of the pa (PulseAudio) audio backend.
+#
+# @server: PulseAudio server address (default: let PulseAudio choose)
+#
+# @sink: name of the sink to use
+#
+# @source: name of the source to use
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevPaOptions',
+ 'data': {
+ '*server': 'str',
+ '*sink': 'AudiodevPaPerDirectionOptions',
+ '*source': 'AudiodevPaPerDirectionOptions' } }
+
+##
+# @AudiodevWavOptions:
+#
+# Options of the wav audio backend.
+#
+# @path: name of the wav file to record (default 'qemu.wav')
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevWavOptions',
+ 'data': {
+ '*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' ] }
+
+##
+# @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 on)
+#
+# @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-len: the buffer size in microseconds
+#
+# @buffer-count: number of buffers
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*fixed-settings': 'bool',
+ '*frequency': 'uint32',
+ '*channels': 'uint32',
+ '*voices': 'uint32',
+ '*format': 'AudioFormat',
+ '*buffer-len': 'uint32',
+ '*buffer-count': 'uint32' } }
+
+##
+# @Audiodev:
+#
+# Options of an audio backend.
+#
+# @id: identifier of the backend
+#
+# @driver: the backend driver to use
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @timer-period: timer period (in microseconds, 0: use lowest possible)
+#
+# Since: 4.0
+##
+{ 'union': 'Audiodev',
+ 'base': {
+ 'id': 'str',
+ 'driver': 'AudiodevDriver',
+ '*in': 'AudiodevPerDirectionOptions',
+ '*out': 'AudiodevPerDirectionOptions',
+ '*timer-period': 'uint32' },
+ 'discriminator': 'driver',
+ 'data': {
+ 'alsa': 'AudiodevAlsaOptions',
+ 'dsound': 'AudiodevDsoundOptions',
+ 'oss': 'AudiodevOssOptions',
+ 'pa': 'AudiodevPaOptions',
+ 'wav': 'AudiodevWavOptions' } }
diff --git a/qapi/qapi-schema.json b/qapi/qapi-schema.json
index 3bbdfcee84..d5f9856be7 100644
--- a/qapi/qapi-schema.json
+++ b/qapi/qapi-schema.json
@@ -95,3 +95,4 @@
{ 'include': 'trace.json' }
{ 'include': 'introspect.json' }
{ 'include': 'misc.json' }
+{ 'include': 'audio.json' }
--
2.20.1
^ permalink raw reply related [flat|nested] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends Kővágó, Zoltán
@ 2019-01-29 13:33 ` Markus Armbruster
2019-02-07 20:26 ` Zoltán Kővágó
2019-01-29 15:52 ` Markus Armbruster
1 sibling, 1 reply; 23+ messages in thread
From: Markus Armbruster @ 2019-01-29 13:33 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: qemu-devel, 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)
>
> AudiodevPaPerDirectionOptions struct might look pointless right now, but
> later we will have new per-direction options, so it's more future-proof
> this way.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
>
> Notes:
> 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
>
> Makefile.objs | 6 +-
> qapi/audio.json | 233 ++++++++++++++++++++++++++++++++++++++++++
> qapi/qapi-schema.json | 1 +
> 3 files changed, 237 insertions(+), 3 deletions(-)
> create mode 100644 qapi/audio.json
>
> diff --git a/Makefile.objs b/Makefile.objs
> index 67a054b08a..d9ee3fed32 100644
> --- a/Makefile.objs
> +++ b/Makefile.objs
> @@ -1,6 +1,6 @@
> -QAPI_MODULES = block-core block char common crypto introspect job migration
> -QAPI_MODULES += misc net rdma rocker run-state sockets tpm trace transaction
> -QAPI_MODULES += ui
> +QAPI_MODULES = audio block-core block char common crypto introspect job
> +QAPI_MODULES += migration misc net rdma rocker run-state sockets tpm trace
> +QAPI_MODULES += transaction ui
>
> #######################################################################
> # Common libraries for tools and emulators
> diff --git a/qapi/audio.json b/qapi/audio.json
> new file mode 100644
> index 0000000000..bd6e2494bd
> --- /dev/null
> +++ b/qapi/audio.json
> @@ -0,0 +1,233 @@
> +# -*- 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.
> +
> +##
> +# @AudiodevAlsaPerDirectionOptions:
> +#
> +# Options of the alsa backend that are used for both playback and recording.
Please break your comment lines around column 70.
> +#
> +# @dev: the name of the alsa device to use (default 'default')
> +#
> +# @try-poll: attempt to use poll mode, falling back to non polling access on
non-polling
> +# failure (default on)
default true
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevAlsaPerDirectionOptions',
> + 'data': {
> + '*dev': 'str',
> + '*try-poll': 'bool' } }
> +
> +##
> +# @AudiodevAlsaOptions:
> +#
> +# Options of the alsa audio backend.
> +#
> +# @alsa-in: options of the capture stream
> +#
> +# @alsa-out: options of the playback stream
> +#
> +# @threshold: set the threshold (in microseconds) when playback starts
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevAlsaOptions',
> + 'data': {
> + '*alsa-in': 'AudiodevAlsaPerDirectionOptions',
> + '*alsa-out': 'AudiodevAlsaPerDirectionOptions',
> + '*threshold': 'uint32' } }
> +
> +##
> +# @AudiodevDsoundOptions:
> +#
> +# Options of the dsound audio backend.
> +#
> +# @latency: add extra latency to playback in microseconds (default 10000)
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevDsoundOptions',
> + 'data': {
> + '*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')
A device special file, unlike AudiodevAlsaPerDirectionOptions member
@dev. Same name, different meaning. Doesn't bother me in this case.
> +#
> +# @try-poll: attempt to use poll mode, falling back to non polling access on
non-polling
> +# failure (default on)
default true
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevOssPerDirectionOptions',
> + 'data': {
> + '*dev': 'str',
> + '*try-poll': 'bool' } }
> +
> +##
> +# @AudiodevOssOptions:
> +#
> +# Options of the oss audio backend.
> +#
> +# @oss-in: options of the capture stream
> +#
> +# @oss-out: options of the playback stream
> +#
> +# @try-mmap: try using memory mapped access, falling back to non
> +# memory mapped access on failure (default off)
memory-mapped, non-memory-mapped
default false
> +#
> +# @exclusive: open device in exclusive mode (vmix won't work) (default off)
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': {
> + '*oss-in': 'AudiodevOssPerDirectionOptions',
> + '*oss-out': 'AudiodevOssPerDirectionOptions',
> + '*try-mmap': 'bool',
> + '*exclusive': 'bool',
> + '*dsp-policy': 'uint32' } }
> +
> +##
> +# @AudiodevPaPerDirectionOptions:
> +#
> +# Options of the pa backend that are used for both playback and recording.
> +#
> +# @name: name of the sink/source to use
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevPaPerDirectionOptions',
> + 'data': {
> + '*name': 'str' } }
Your "might look pointless" remark made me expect an empty struct. That
would've been pointless indeed. Instead, you're wrapping the single
option you already have in a struct, to keep open the door for more
options. Makes sense.
> +
> +##
> +# @AudiodevPaOptions:
> +#
> +# Options of the pa (PulseAudio) audio backend.
> +#
> +# @server: PulseAudio server address (default: let PulseAudio choose)
> +#
> +# @sink: name of the sink to use
> +#
> +# @source: name of the source to use
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevPaOptions',
> + 'data': {
> + '*server': 'str',
> + '*sink': 'AudiodevPaPerDirectionOptions',
> + '*source': 'AudiodevPaPerDirectionOptions' } }
> +
> +##
> +# @AudiodevWavOptions:
> +#
> +# Options of the wav audio backend.
> +#
> +# @path: name of the wav file to record (default 'qemu.wav')
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevWavOptions',
> + 'data': {
> + '*path': 'str' } }
Pattern so far:
* AudiodevFooOptions
Foo direction options other options
Alsa alsa-in, alsa-out threshold
Dsound latency
Oss oss-in, oss-out try-map, exclusive, dsp-policy
Pa sink, source server
Wav path
Any particular reason for naming the direction options differently?
> +
> +
> +##
> +# @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' ] }
> +
> +##
> +# @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 on)
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-len: the buffer size in microseconds
> +#
> +# @buffer-count: number of buffers
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevPerDirectionOptions',
> + 'data': {
> + '*fixed-settings': 'bool',
> + '*frequency': 'uint32',
> + '*channels': 'uint32',
> + '*voices': 'uint32',
> + '*format': 'AudioFormat',
> + '*buffer-len': 'uint32',
> + '*buffer-count': 'uint32' } }
> +
> +##
> +# @Audiodev:
> +#
> +# Options of an audio backend.
> +#
> +# @id: identifier of the backend
> +#
> +# @driver: the backend driver to use
> +#
> +# @in: options of the capture stream
> +#
> +# @out: options of the playback stream
> +#
> +# @timer-period: timer period (in microseconds, 0: use lowest possible)
> +#
> +# Since: 4.0
> +##
> +{ 'union': 'Audiodev',
> + 'base': {
> + 'id': 'str',
> + 'driver': 'AudiodevDriver',
> + '*in': 'AudiodevPerDirectionOptions',
> + '*out': 'AudiodevPerDirectionOptions',
> + '*timer-period': 'uint32' },
> + 'discriminator': 'driver',
> + 'data': {
> + 'alsa': 'AudiodevAlsaOptions',
> + 'dsound': 'AudiodevDsoundOptions',
> + 'oss': 'AudiodevOssOptions',
> + 'pa': 'AudiodevPaOptions',
> + 'wav': 'AudiodevWavOptions' } }
'none', 'coreaudio', 'sdl' and 'spice' have no driver-specific options.
Do all the generic options apply to all drivers? Even 'none'?
Do @in and @out apply to drivers 'dsound' and 'wav'?
Per-direction options are split between generic ones in @in and @out,
and driver-specific ones in @alsa-in and @alsa-out / @oss-in and
@oss-out / @sink and @source.
Perhaps the following would be tidier: make the generic per-direction
options AudiodevPerDirectionOptions the base of driver-specific ones
AudiodevAlsaPerDirectionOptions, AudiodevOssPerDirectionOptions,
AudiodevPaPerDirectionOptions, and then
{ 'struct': 'AudiodevGenericOptions',
'data': {
'*in': 'AudiodevPerDirectionOptions',
'*out': 'AudiodevPerDirectionOptions' } }
{ 'union': 'Audiodev',
'base': {
'id': 'str',
'driver': 'AudiodevDriver',
'*timer-period': 'uint32' },
'discriminator': 'driver',
'data': {
'none': 'AudiodevGenericOptions',
'alsa': 'AudiodevAlsaOptions',
'coreaudio: 'AudiodevGenericOptions',
'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 3bbdfcee84..d5f9856be7 100644
> --- a/qapi/qapi-schema.json
> +++ b/qapi/qapi-schema.json
> @@ -95,3 +95,4 @@
> { 'include': 'trace.json' }
> { 'include': 'introspect.json' }
> { 'include': 'misc.json' }
> +{ 'include': 'audio.json' }
Looks pretty good.
^ permalink raw reply [flat|nested] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends
2019-01-29 13:33 ` Markus Armbruster
@ 2019-02-07 20:26 ` Zoltán Kővágó
0 siblings, 0 replies; 23+ messages in thread
From: Zoltán Kővágó @ 2019-02-07 20:26 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Gerd Hoffmann
On 2019-01-29 14:33, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
[...]
>> +
>> +##
>> +# @AudiodevPaOptions:
>> +#
>> +# Options of the pa (PulseAudio) audio backend.
>> +#
>> +# @server: PulseAudio server address (default: let PulseAudio choose)
>> +#
>> +# @sink: name of the sink to use
>> +#
>> +# @source: name of the source to use
>> +#
>> +# Since: 4.0
>> +##
>> +{ 'struct': 'AudiodevPaOptions',
>> + 'data': {
>> + '*server': 'str',
>> + '*sink': 'AudiodevPaPerDirectionOptions',
>> + '*source': 'AudiodevPaPerDirectionOptions' } }
>> +
>> +##
>> +# @AudiodevWavOptions:
>> +#
>> +# Options of the wav audio backend.
>> +#
>> +# @path: name of the wav file to record (default 'qemu.wav')
>> +#
>> +# Since: 4.0
>> +##
>> +{ 'struct': 'AudiodevWavOptions',
>> + 'data': {
>> + '*path': 'str' } }
>
> Pattern so far:
>
> * AudiodevFooOptions
>
> Foo direction options other options
> Alsa alsa-in, alsa-out threshold
> Dsound latency
> Oss oss-in, oss-out try-map, exclusive, dsp-policy
> Pa sink, source server
> Wav path
>
> Any particular reason for naming the direction options differently?
I probably chose that name because in PulseAudio parlance you have
sources and sinks, and the current environment variable based setting of
pa doesn't use ADC/DAC to refer to source/sink, but now that you mention
it it probably makes sense to rename them to pa-in and pa-out, it's more
uniform that way.
[...]
>> +##
>> +# @Audiodev:
>> +#
>> +# Options of an audio backend.
>> +#
>> +# @id: identifier of the backend
>> +#
>> +# @driver: the backend driver to use
>> +#
>> +# @in: options of the capture stream
>> +#
>> +# @out: options of the playback stream
>> +#
>> +# @timer-period: timer period (in microseconds, 0: use lowest possible)
>> +#
>> +# Since: 4.0
>> +##
>> +{ 'union': 'Audiodev',
>> + 'base': {
>> + 'id': 'str',
>> + 'driver': 'AudiodevDriver',
>> + '*in': 'AudiodevPerDirectionOptions',
>> + '*out': 'AudiodevPerDirectionOptions',
>> + '*timer-period': 'uint32' },
>> + 'discriminator': 'driver',
>> + 'data': {
>> + 'alsa': 'AudiodevAlsaOptions',
>> + 'dsound': 'AudiodevDsoundOptions',
>> + 'oss': 'AudiodevOssOptions',
>> + 'pa': 'AudiodevPaOptions',
>> + 'wav': 'AudiodevWavOptions' } }
>
> 'none', 'coreaudio', 'sdl' and 'spice' have no driver-specific options.
>
> Do all the generic options apply to all drivers? Even 'none'?
'fixed-settings', 'frequency', 'channels', 'voices' and 'format' are
used by the mixeng thus they apply to all backends, even 'none'.
(Whether it makes any sense to set them is a different story.)
'buffer-len' and 'buffer-count' was introduced as a means to give a more
uniform interface to setting buffer sizes, because currently to set
buffer size:
* in case of alsa: you set microseconds or frames depending on a second
environment variable
* in case of coreaudio: you set it in frames
* in case of dsound, oss: you set it in milliseconds
* in case of pa, sdl: you set it in samples
* in case of spice, wav: you can't set it
I added 'buffer-len' and 'buffer-count' because they seemed like a
generic option that *most* backends would support, and I didn't want to
create extra backend specific options for them. But if I combine it
with your inheritance-like proposal, it's probably not that bad.
> Do @in and @out apply to drivers 'dsound' and 'wav'?
@out applies to every driver. @in shouldn't apply to drivers without
input support, but currently the base of mixeng is created for the input
too, it'll just fail if the sound card tries to create a capture stream...
>
> Per-direction options are split between generic ones in @in and @out,
> and driver-specific ones in @alsa-in and @alsa-out / @oss-in and
> @oss-out / @sink and @source.
>
> Perhaps the following would be tidier: make the generic per-direction
> options AudiodevPerDirectionOptions the base of driver-specific ones
> AudiodevAlsaPerDirectionOptions, AudiodevOssPerDirectionOptions,
> AudiodevPaPerDirectionOptions, and then
>
> { 'struct': 'AudiodevGenericOptions',
> 'data': {
> '*in': 'AudiodevPerDirectionOptions',
> '*out': 'AudiodevPerDirectionOptions' } }
>
> { 'union': 'Audiodev',
> 'base': {
> 'id': 'str',
> 'driver': 'AudiodevDriver',
> '*timer-period': 'uint32' },
> 'discriminator': 'driver',
> 'data': {
> 'none': 'AudiodevGenericOptions',
> 'alsa': 'AudiodevAlsaOptions',
> 'coreaudio: 'AudiodevGenericOptions',
> 'dsound': 'AudiodevDsoundOptions',
> 'oss': 'AudiodevOssOptions',
> 'pa': 'AudiodevPaOptions',
> 'sdl: 'AudiodevGenericOptions',
> 'spice': 'AudiodevGenericOptions',
> 'wav': 'AudiodevWavOptions' } }
Neat, I'll play around with this a bit.
>> diff --git a/qapi/qapi-schema.json b/qapi/qapi-schema.json
>> index 3bbdfcee84..d5f9856be7 100644
>> --- a/qapi/qapi-schema.json
>> +++ b/qapi/qapi-schema.json
>> @@ -95,3 +95,4 @@
>> { 'include': 'trace.json' }
>> { 'include': 'introspect.json' }
>> { 'include': 'misc.json' }
>> +{ 'include': 'audio.json' }
>
> Looks pretty good.
>
Regards,
Zoltan
^ permalink raw reply [flat|nested] 23+ messages in thread
* Re: [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends Kővágó, Zoltán
2019-01-29 13:33 ` Markus Armbruster
@ 2019-01-29 15:52 ` Markus Armbruster
1 sibling, 0 replies; 23+ messages in thread
From: Markus Armbruster @ 2019-01-29 15:52 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: qemu-devel, 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)
>
> AudiodevPaPerDirectionOptions struct might look pointless right now, but
> later we will have new per-direction options, so it's more future-proof
> this way.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
[...]
> diff --git a/qapi/audio.json b/qapi/audio.json
> new file mode 100644
> index 0000000000..bd6e2494bd
> --- /dev/null
> +++ b/qapi/audio.json
[...]
> +##
> +# @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 on)
> +#
> +# @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)
If I specify any of @frequency, @channels, @voices, @format, then I have
to have "fixed-settings": true. Since true is the default, I don't have
to say it explicitly, which would be redundant and mildly annoying.
However, specifying "fixed-settings": false is *not* redundant, because
it's different from specifying nothing at all, which is the same as
"fixed-settings": true, "frequency": 44100, "channels": 2, "voices": 1,
"format": "s16".
I think I get it.
> +#
> +# @buffer-len: the buffer size in microseconds
> +#
> +# @buffer-count: number of buffers
> +#
> +# Since: 4.0
> +##
[...]
^ permalink raw reply [flat|nested] 23+ messages in thread
* [Qemu-devel] [PATCH v4 02/14] audio: use qapi AudioFormat instead of audfmt_e
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-29 6:22 ` Thomas Huth
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 03/14] audio: -audiodev command line option: documentation Kővágó, Zoltán
` (12 subsequent siblings)
14 siblings, 1 reply; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel
Cc: Gerd Hoffmann, 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>
---
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 d163ffbc88..ce8e6ea8c2 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -113,7 +113,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -125,7 +125,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s)
return r;
}
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return "U8";
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return "U16";
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return "S8";
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return "S16";
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
return "U32";
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
return "S32";
+
+ default:
+ abort();
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
-static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
- return AUD_FMT_U16;
+ return AUDIO_FORMAT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
- return AUD_FMT_U32;
+ return AUDIO_FORMAT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
- return AUD_FMT_S8;
+ return AUDIO_FORMAT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
@@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
}
}
-static audfmt_e audio_get_conf_fmt (const char *envname,
- audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+ AudioFormat defval,
int *defaultp)
{
const char *var = getenv (envname);
@@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
@@ -508,7 +511,7 @@ static void audio_process_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
@@ -539,22 +542,22 @@ static void audio_print_settings (struct audsettings *as)
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
AUD_log (NULL, "S8");
break;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
AUD_log (NULL, "U8");
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
AUD_log (NULL, "S16");
break;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
AUD_log (NULL, "U16");
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
AUD_log (NULL, "S32");
break;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
default:
@@ -585,12 +588,12 @@ static int audio_validate_settings (struct audsettings *as)
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
break;
default:
invalid = 1;
@@ -606,25 +609,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
int bits = 8, sign = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
break;
+
+ default:
+ abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
@@ -638,24 +644,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
+
+ default:
+ abort();
}
info->freq = as->freq;
diff --git a/audio/audio_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 9db5ac92bc..aa42ea26bf 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -137,19 +137,19 @@ static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
return sdl_post (s, forfn);
}
-static int aud_to_sdlfmt (audfmt_e fmt)
+static int aud_to_sdlfmt (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AUDIO_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AUDIO_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return AUDIO_S16LSB;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return AUDIO_U16LSB;
default:
@@ -161,37 +161,37 @@ static int aud_to_sdlfmt (audfmt_e fmt)
}
}
-static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
+static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
{
switch (sdlfmt) {
case AUDIO_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AUDIO_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AUDIO_S16LSB:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16LSB:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AUDIO_S16MSB:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16MSB:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -386,7 +386,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
SDL_AudioSpec req, obt;
int endianness;
int err;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
req.freq = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6ad0eafbc6..3aeb0cb357 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
settings.freq = SPICE_INTERFACE_PLAYBACK_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
@@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.freq = SPICE_INTERFACE_RECORD_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 40adfa30c3..35a614785e 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
bits16 = 0;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bits16 = 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
+
+ default:
+ abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
@@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
+ .settings.fmt = AUDIO_FORMAT_S16,
.wav_path = "qemu.wav"
};
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index cf31ed652c..e943565210 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -139,7 +139,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
as.freq = freq;
as.nchannels = 1 << stereo;
- as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
ops.notify = wav_notify;
diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c
index 3c7d1364a9..96e84f6e7e 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -273,7 +273,7 @@ static void omap_eac_format_update(struct omap_eac_s *s)
* does I2S specify it? */
/* All register writes are 16 bits so we we store 16-bit samples
* in the buffers regardless of AGCFR[B8_16] value. */
- fmt.fmt = AUD_FMT_U16;
+ fmt.fmt = AUDIO_FORMAT_U16;
s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
"eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index d799533aa9..2265622d44 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 97b876c7e0..0957780a3d 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 9089dcb47e..62da75eefe 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
case 0:
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
s->shift = as.nchannels == 2;
break;
@@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 3:
s->tab = ALawDecompressTable;
x_law:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
s->shift = as.nchannels == 2;
break;
@@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
as.endianness = 1;
/* fall through */
case 2:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
s->shift = as.nchannels;
break;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 97789a0771..a5314d66fd 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
i,
new_freq,
1 << (new_fmt & 1),
- (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+ (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8,
d->shift);
if (new_freq) {
struct audsettings as;
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
- as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
if (i == ADC_CHANNEL) {
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 8e0b27e0f2..b3e2a7fdd5 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = GUS_ENDIANNESS;
s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 617a1c1016..c25bfa38b1 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
}
switch (format & AC_FMT_BITS_MASK) {
- case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
- case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
- case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
- [ AUD_FMT_U8 ] = "PCM-U8",
- [ AUD_FMT_S8 ] = "PCM-S8",
- [ AUD_FMT_U16 ] = "PCM-U16",
- [ AUD_FMT_S16 ] = "PCM-S16",
- [ AUD_FMT_U32 ] = "PCM-U32",
- [ AUD_FMT_S32 ] = "PCM-S32",
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index a46f2301af..af8b22b541 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s,
struct audsettings as;
as.freq = value;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id)
struct audsettings as;
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque)
/* Open a default voice */
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index bc8db71ae0..90cce1e6ed 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp)
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 1;
s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 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 f4aa838f62..d563f31e7c 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 ded0db9351..885f71366c 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 6002d09407..2189f0c080 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;
}
@@ -2428,12 +2428,12 @@ 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);
@@ -3164,7 +3164,7 @@ 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] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 02/14] audio: use qapi AudioFormat instead of audfmt_e
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 02/14] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
@ 2019-01-29 6:22 ` Thomas Huth
0 siblings, 0 replies; 23+ messages in thread
From: Thomas Huth @ 2019-01-29 6:22 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel
Cc: Peter Maydell, Alistair Francis, Michael Walle, open list:ARM,
Gerd Hoffmann, Edgar E. Iglesias
On 2019-01-28 23:43, Kővágó, Zoltán wrote:
> 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>
^ permalink raw reply [flat|nested] 23+ messages in thread
* [Qemu-devel] [PATCH v4 03/14] audio: -audiodev command line option: documentation
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 01/14] qapi: qapi for audio backends Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 02/14] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-29 6:13 ` Thomas Huth
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 04/14] audio: -audiodev command line option basic implementation Kővágó, Zoltán
` (11 subsequent siblings)
14 siblings, 1 reply; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This patch adds documentation of an -audiodev command line option, that
deprecates the old QEMU_* environment variables for audio backend
configuration. It's syntax is similar to existing options (-netdev,
-device, etc):
-audiodev driver_name,property=value,...
Although now it's possible to specify multiple -audiodev options on
command line, multiple audio backends are not supported yet.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
qemu-options.hx | 222 +++++++++++++++++++++++++++++++++++++++++++++++-
1 file changed, 219 insertions(+), 3 deletions(-)
diff --git a/qemu-options.hx b/qemu-options.hx
index 521511ec13..a12931899b 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -416,14 +416,230 @@ The default is @code{en-us}.
ETEXI
+HXCOMM Deprecated by -audiodev
DEF("audio-help", 0, QEMU_OPTION_audio_help,
- "-audio-help print list of audio drivers and their options\n",
+ "-audio-help show -audiodev equivalent of the currently specified audio settings\n",
QEMU_ARCH_ALL)
STEXI
@item -audio-help
@findex -audio-help
-Will show the audio subsystem help: list of drivers, tunable
-parameters.
+Will show the -audiodev equivalent of the currently specified
+(deprecated) environment variables.
+ETEXI
+
+DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
+ "-audiodev [driver=]driver,id=id[,prop[=value][,...]]\n"
+ " specifies the audio backend to use\n"
+ " id= identifier of the backend\n"
+ " timer-period= timer period in microseconds\n"
+ " in|out.fixed-settings= use fixed settings for host audio\n"
+ " in|out.frequency= frequency to use with fixed settings\n"
+ " in|out.channels= number of channels to use with fixed settings\n"
+ " in|out.format= sample format to use with fixed settings\n"
+ " valid values: s8, s16, s32, u8, u16, u32\n"
+ " in|out.voices= number of voices to use\n"
+ " in|out.buffer-len= size of buffer in microseconds\n"
+ " in|out.buffer-count= number of buffers\n"
+ "-audiodev none,id=id,[,prop[=value][,...]]\n"
+ " dummy driver that discards all output\n"
+#ifdef CONFIG_ALSA
+ "-audiodev alsa,id=id[,prop[=value][,...]]\n"
+ " alsa-in|alsa-out.dev= name of the audio device to use\n"
+ " alsa-in|alsa-out.try-poll= attempt to use poll mode\n"
+ " threshold= threshold (in microseconds) when playback starts\n"
+#endif
+#ifdef CONFIG_COREAUDIO
+ "-audiodev coreaudio,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_DSOUND
+ "-audiodev dsound,id=id[,prop[=value][,...]]\n"
+ " latency= add extra latency to playback in microseconds\n"
+#endif
+#ifdef CONFIG_OSS
+ "-audiodev oss,id=id[,prop[=value][,...]]\n"
+ " oss-in|oss-out.dev= path of the audio device to use\n"
+ " oss-in|oss-out.try-poll= attempt to use poll mode\n"
+ " try-mmap= try using memory mapped access\n"
+ " exclusive= open device in exclusive mode\n"
+ " dsp-policy= set timing policy (0..10), -1 to use fragment mode\n"
+#endif
+#ifdef CONFIG_PA
+ "-audiodev pa,id=id[,prop[=value][,...]]\n"
+ " server= PulseAudio server address\n"
+ " sink|source.name= sink/source device name\n"
+#endif
+#ifdef CONFIG_SDL
+ "-audiodev sdl,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_SPICE
+ "-audiodev spice,id=id[,prop[=value][,...]]\n"
+#endif
+ "-audiodev wav,id=id[,prop[=value][,...]]\n"
+ " path= path of wav file to record\n",
+ QEMU_ARCH_ALL)
+STEXI
+@item -audiodev [driver=]@var{driver},id=@var{id}[,@var{prop}[=@var{value}][,...]]
+@findex -audiodev
+Adds a new audio backend @var{driver} identified by @var{id}. There are
+global and driver specific properties. Some values can be set
+differently for input and output, they're marked with @code{in|out.}.
+You can set the input's property with @code{in.@var{prop}} and the
+output's property with @code{out.@var{prop}}. For example:
+@example
+-audiodev alsa,id=example,in.frequency=44110,out.frequency=8000
+-audiodev alsa,id=example,out.channels=1 # leaves in.channels unspecified
+@end example
+
+Valid global options are:
+
+@table @option
+@item id=@var{identifier}
+Identifies the audio backend.
+
+@item timer-period=@var{period}
+Sets the timer @var{period} used by the audio subsystem in microseconds.
+Default is 10000 (10 ms).
+
+@item in|out.fixed-settings=on|off
+Use fixed settings for host audio. When off, it will change based on
+how the guest opens the sound card. In this case you must not specify
+@var{frequency}, @var{channels} or @var{format}. Default is on.
+
+@item in|out.frequency=@var{frequency}
+Specify the @var{frequency} to use when using @var{fixed-settings}.
+Default is 44100Hz.
+
+@item in|out.channels=@var{channels}
+Specify the number of @var{channels} to use when using
+@var{fixed-settings}. Default is 2 (stereo).
+
+@item in|out.format=@var{format}
+Specify the sample @var{format} to use when using @var{fixed-settings}.
+Valid values are: @code{s8}, @code{s16}, @code{s32}, @code{u8},
+@code{u16}, @code{u32}. Default is @code{s16}.
+
+@item in|out.voices=@var{voices}
+Specify the number of @var{voices} to use. Default is 1.
+
+@item in|out.buffer=@var{usecs}
+Sets the size of the buffer in microseconds.
+
+@item in|out.buffer-count=@var{count}
+Sets the @var{count} of the buffers.
+
+@end table
+
+@item -audiodev none,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a dummy backend that discards all outputs. This backend has no
+backend specific properties.
+
+@item -audiodev alsa,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates backend using the ALSA. This backend is only available on
+Linux.
+
+ALSA specific options are:
+
+@table @option
+@item alsa-in|alsa-out.dev=@var{device}
+Specify the ALSA @var{device} to use for input and/or output. Default
+is @code{default}.
+
+@item alsa-in|alsa-out.try-poll=on|off
+Attempt to use poll mode with the device. Default is on.
+
+@item threshold=@var{threshold}
+Threshold (in microseconds) when playback starts. Default is 0.
+
+@end table
+
+@item -audiodev coreaudio,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using Apple's Core Audio. This backend is only
+available on Mac OS and only supports playback. This backend has no
+backend specific properties.
+
+@item -audiodev dsound,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using Microsoft's DirectSound. This backend is only
+available on Windows and only supports playback.
+
+Backend specific options are:
+
+@table @option
+
+@item latency=@var{usecs}
+Add extra @var{usecs} microseconds latency to playback. Default is
+10000 (10 ms).
+
+@end table
+
+@item -audiodev oss,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using OSS. This backend is available on most
+Unix-like systems.
+
+OSS specific options are:
+
+@table @option
+
+@item oss-in|oss-out.dev=@var{device}
+Specify the file name of the OSS @var{device} to use. Default is
+@code{/dev/dsp}.
+
+@item oss-in|oss-out.try-poll=on|of
+Attempt to use poll mode with the device. Default is on.
+
+@item try-mmap=on|off
+Try using memory mapped device access. Default is off.
+
+@item exclusive=on|off
+Open the device in exclusive mode (vmix won't work in this case).
+Default is off.
+
+@item dsp-policy=@var{policy}
+Sets the timing policy (between 0 and 10, where smaller number means
+smaller latency but higher CPU usage). Use -1 to use buffer sizes
+specified by @code{buffer} and @code{buffer-count}. This option is
+ignored if you do not have OSS 4. Default is 5.
+
+@end table
+
+@item -audiodev pa,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using PulseAudio. This backend is available on most
+systems.
+
+PulseAudio specific options are:
+
+@table @option
+
+@item server=@var{server}
+Sets the PulseAudio @var{server} to connect to.
+
+@item sink|source.name=@var{sink}
+Use the specified sink/source for playback/recording.
+
+@end table
+
+@item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using SDL. This backend is available on most systems,
+but you should use your platform's native backend if possible. This
+backend has no backend specific properties.
+
+@item -audiodev spice,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend that sends audio through SPICE. This backend requires
+@code{-spice} and automatically selected in that case, so usually you
+can ignore this option. This backend has no backend specific
+properties.
+
+@item -audiodev wav,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend that writes audio to a WAV file.
+
+Backend specific options are:
+
+@table @option
+
+@item path=@var{path}
+Write recorded audio into the specified file. Default is
+@code{qemu.wav}.
+
+@end table
ETEXI
DEF("soundhw", HAS_ARG, QEMU_OPTION_soundhw,
--
2.20.1
^ permalink raw reply related [flat|nested] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 03/14] audio: -audiodev command line option: documentation
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 03/14] audio: -audiodev command line option: documentation Kővágó, Zoltán
@ 2019-01-29 6:13 ` Thomas Huth
0 siblings, 0 replies; 23+ messages in thread
From: Thomas Huth @ 2019-01-29 6:13 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel; +Cc: Gerd Hoffmann, Libvirt
On 2019-01-28 23:43, Kővágó, Zoltán wrote:
> This patch adds documentation of an -audiodev command line option, that
> deprecates the old QEMU_* environment variables for audio backend
> configuration. It's syntax is similar to existing options (-netdev,
> -device, etc):
>
> -audiodev driver_name,property=value,...
>
> Although now it's possible to specify multiple -audiodev options on
> command line, multiple audio backends are not supported yet.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> qemu-options.hx | 222 +++++++++++++++++++++++++++++++++++++++++++++++-
> 1 file changed, 219 insertions(+), 3 deletions(-)
>
> diff --git a/qemu-options.hx b/qemu-options.hx
> index 521511ec13..a12931899b 100644
> --- a/qemu-options.hx
> +++ b/qemu-options.hx
> @@ -416,14 +416,230 @@ The default is @code{en-us}.
> ETEXI
>
>
> +HXCOMM Deprecated by -audiodev
> DEF("audio-help", 0, QEMU_OPTION_audio_help,
> - "-audio-help print list of audio drivers and their options\n",
> + "-audio-help show -audiodev equivalent of the currently specified audio settings\n",
> QEMU_ARCH_ALL)
Note that if you want to deprecate a parameter "officially" (in the
sense that it could be removed in a future release), you also have to
add an entry to qemu-deprecated.texi. But that could also be done in an
additional patch later, so no need to respin just because of this.
Thomas
^ permalink raw reply [flat|nested] 23+ messages in thread
* [Qemu-devel] [PATCH v4 04/14] audio: -audiodev command line option basic implementation
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (2 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 03/14] audio: -audiodev command line option: documentation Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-29 15:56 ` Markus Armbruster
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 05/14] alsaaudio: port to -audiodev config Kővágó, Zoltán
` (10 subsequent siblings)
14 siblings, 1 reply; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 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>
---
Notes:
Changes from v2:
* MAJOR: use qobject_input_visitor instead of QemuOpts
* almost completely rewrote legacy options handling
* added missing license comment to audio_legacy.c0
audio/audio.h | 18 +-
audio/audio_int.h | 16 +-
audio/audio_template.h | 13 +-
audio/alsaaudio.c | 2 +-
audio/audio.c | 601 ++++++++++++++++-------------------------
audio/audio_legacy.c | 295 ++++++++++++++++++++
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, 590 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..cee46c4809 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 +252,14 @@ 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);
+
#endif /* QEMU_AUDIO_INT_H */
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 7de227d2d1..c1d7207abd 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -302,8 +302,10 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
{
HW *hw;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
- if (glue (conf.fixed_, TYPE).enabled && glue (conf.fixed_, TYPE).greedy) {
+ if (pdo->fixed_settings) {
hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
if (hw) {
return hw;
@@ -331,9 +333,11 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
SW *sw;
HW *hw;
struct audsettings hw_as;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
- if (glue (conf.fixed_, TYPE).enabled) {
- hw_as = glue (conf.fixed_, TYPE).settings;
+ if (pdo->fixed_settings) {
+ hw_as = audiodev_to_audsettings(pdo);
}
else {
hw_as = *as;
@@ -398,6 +402,7 @@ SW *glue (AUD_open_, TYPE) (
)
{
AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -422,7 +427,7 @@ SW *glue (AUD_open_, TYPE) (
return sw;
}
- if (!glue (conf.fixed_, TYPE).enabled && sw) {
+ if (!pdo->fixed_settings && sw) {
glue (AUD_close_, TYPE) (card, sw);
sw = NULL;
}
diff --git a/audio/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 ce8e6ea8c2..b37c245a8a 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)
{
@@ -1161,11 +1004,11 @@ static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed ()) {
timer_mod_anticipate_ns(s->ts,
- qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+ qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
if (!audio_timer_running) {
audio_timer_running = true;
audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- trace_audio_timer_start(conf.period.ticks / SCALE_MS);
+ trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
@@ -1179,16 +1022,17 @@ static void audio_reset_timer (AudioState *s)
static void audio_timer (void *opaque)
{
int64_t now, diff;
+ AudioState *s = opaque;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
diff = now - audio_timer_last;
- if (diff > conf.period.ticks * 3 / 2) {
+ if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
audio_timer_last = now;
- audio_run ("timer");
- audio_reset_timer (opaque);
+ audio_run("timer");
+ audio_reset_timer(s);
}
/*
@@ -1248,7 +1092,7 @@ 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);
}
}
@@ -1293,7 +1137,7 @@ 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);
}
}
@@ -1614,169 +1458,13 @@ 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);
@@ -1802,11 +1490,11 @@ 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);
}
@@ -1856,6 +1544,11 @@ 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 = {
@@ -1867,19 +1560,58 @@ 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);
@@ -1887,10 +1619,8 @@ static void audio_init (void)
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
- audio_process_options ("AUDIO", audio_options);
-
- s->nb_hw_voices_out = conf.fixed_out.nb_voices;
- s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+ s->nb_hw_voices_out = dev->out->voices;
+ s->nb_hw_voices_in = dev->in->voices;
if (s->nb_hw_voices_out <= 0) {
dolog ("Bogus number of playback voices %d, setting to 1\n",
@@ -1904,46 +1634,42 @@ 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);
@@ -1954,11 +1680,22 @@ 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);
@@ -2129,3 +1866,145 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
}
}
}
+
+static void audio_validate_per_direction_opts(
+ AudiodevPerDirectionOptions **pdo, bool *has_pdo, Error **errp)
+{
+ if (!*has_pdo) {
+ *pdo = g_malloc0(sizeof(AudiodevPerDirectionOptions));
+ *has_pdo = true;
+ }
+
+ 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_validate_per_direction_opts(&dev->in, &dev->has_in, &err);
+ if (err) {
+ error_propagate(errp, err);
+ return;
+ }
+
+ audio_validate_per_direction_opts(&dev->out, &dev->has_out, &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_len ? pdo->buffer_len : def_usecs;
+ return (as->freq * usecs + 500000) / 1000000;
+}
+
+/* samples = channels * frames = channels * freq * usec / 1e6 */
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
+}
+
+/*
+ * bytes = bytes_per_sample * samples =
+ * bytes_per_sample * channels * freq * usec / 1e6
+ */
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return audio_buffer_samples(pdo, as, def_usecs) *
+ audioformat_bytes_per_sample(as->fmt);
+}
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
new file mode 100644
index 0000000000..c0b2c4604e
--- /dev/null
+++ b/audio/audio_legacy.c
@@ -0,0 +1,295 @@
+/*
+ * 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);
+ e->dev->in = g_malloc0(sizeof(AudiodevPerDirectionOptions));
+ e->dev->has_in = true;
+ e->dev->out = g_malloc0(sizeof(AudiodevPerDirectionOptions));
+ e->dev->has_out = true;
+
+ handle_per_direction(e->dev->in, "QEMU_AUDIO_ADC_");
+ handle_per_direction(e->dev->out, "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 aa42ea26bf..097841fde1 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -436,7 +436,7 @@ static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static void *sdl_audio_init (void)
+static void *sdl_audio_init(Audiodev *dev)
{
SDLAudioState *s = &glob_sdl;
if (s->driver_created) {
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 3aeb0cb357..affc3df17f 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -77,7 +77,7 @@ static const SpiceRecordInterface record_sif = {
.base.minor_version = SPICE_INTERFACE_RECORD_MINOR,
};
-static void *spice_audio_init (void)
+static void *spice_audio_init(Audiodev *dev)
{
if (!using_spice) {
return NULL;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 35a614785e..9eff3555b3 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -232,7 +232,7 @@ static WAVConf glob_conf = {
.wav_path = "qemu.wav"
};
-static void *wav_audio_init (void)
+static void *wav_audio_init(Audiodev *dev)
{
WAVConf *conf = g_malloc(sizeof(WAVConf));
*conf = glob_conf;
diff --git a/vl.c b/vl.c
index bc9fbec654..80aa0458fc 100644
--- a/vl.c
+++ b/vl.c
@@ -3302,9 +3302,12 @@ 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;
@@ -4533,6 +4536,8 @@ 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] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 04/14] audio: -audiodev command line option basic implementation
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 04/14] audio: -audiodev command line option basic implementation Kővágó, Zoltán
@ 2019-01-29 15:56 ` Markus Armbruster
2019-02-07 20:29 ` Zoltán Kővágó
0 siblings, 1 reply; 23+ messages in thread
From: Markus Armbruster @ 2019-01-29 15:56 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: qemu-devel, Paolo Bonzini, Gerd Hoffmann
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> Audio drivers now get an Audiodev * as config paramters, instead of the
> global audio_option structs. There is some code in audio/audio_legacy.c
> that converts the old environment variables to audiodev options (this
> way backends do not have to worry about legacy options). It also
> contains a replacement of -audio-help, which prints out the equivalent
> -audiodev based config of the currently specified environment variables.
>
> Note that backends are not updated and still rely on environment
> variables.
>
> Also note that (due to moving try-poll from global to backend specific
> option) currently ALSA and OSS will always try poll mode, regardless of
> environment variables or -audiodev options.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
[...]
> diff --git a/audio/audio.c b/audio/audio.c
> index ce8e6ea8c2..b37c245a8a 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
[...]
> @@ -2129,3 +1866,145 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
> }
> }
> }
> +
> +static void audio_validate_per_direction_opts(
> + AudiodevPerDirectionOptions **pdo, bool *has_pdo, Error **errp)
> +{
> + if (!*has_pdo) {
> + *pdo = g_malloc0(sizeof(AudiodevPerDirectionOptions));
> + *has_pdo = true;
> + }
> +
> + 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_validate_per_direction_opts(&dev->in, &dev->has_in, &err);
> + if (err) {
> + error_propagate(errp, err);
> + return;
> + }
> +
> + audio_validate_per_direction_opts(&dev->out, &dev->has_out, &err);
> + if (err) {
> + error_propagate(errp, err);
> + return;
> + }
> +
> + if (!dev->has_timer_period) {
> + dev->has_timer_period = true;
> + dev->timer_period = 10000; /* 100Hz -> 10ms */
> + }
> +}
Drive-by question: this validates just the generic part of Audiodev gets
validated, not the driver-specific part. Is there nothing to validate?
Does it happen somewhere else?
> +
> +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);
> + }
> +}
Your use of QAPI here looks good to me.
[...]
^ permalink raw reply [flat|nested] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 04/14] audio: -audiodev command line option basic implementation
2019-01-29 15:56 ` Markus Armbruster
@ 2019-02-07 20:29 ` Zoltán Kővágó
0 siblings, 0 replies; 23+ messages in thread
From: Zoltán Kővágó @ 2019-02-07 20:29 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Paolo Bonzini, Gerd Hoffmann
On 2019-01-29 16:56, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> Audio drivers now get an Audiodev * as config paramters, instead of the
>> global audio_option structs. There is some code in audio/audio_legacy.c
>> that converts the old environment variables to audiodev options (this
>> way backends do not have to worry about legacy options). It also
>> contains a replacement of -audio-help, which prints out the equivalent
>> -audiodev based config of the currently specified environment variables.
>>
>> Note that backends are not updated and still rely on environment
>> variables.
>>
>> Also note that (due to moving try-poll from global to backend specific
>> option) currently ALSA and OSS will always try poll mode, regardless of
>> environment variables or -audiodev options.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
> [...]
>> diff --git a/audio/audio.c b/audio/audio.c
>> index ce8e6ea8c2..b37c245a8a 100644
>> --- a/audio/audio.c
>> +++ b/audio/audio.c
> [...]
>> @@ -2129,3 +1866,145 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
>> }
>> }
>> }
>> +
>> +static void audio_validate_per_direction_opts(
>> + AudiodevPerDirectionOptions **pdo, bool *has_pdo, Error **errp)
>> +{
>> + if (!*has_pdo) {
>> + *pdo = g_malloc0(sizeof(AudiodevPerDirectionOptions));
>> + *has_pdo = true;
>> + }
>> +
>> + 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_validate_per_direction_opts(&dev->in, &dev->has_in, &err);
>> + if (err) {
>> + error_propagate(errp, err);
>> + return;
>> + }
>> +
>> + audio_validate_per_direction_opts(&dev->out, &dev->has_out, &err);
>> + if (err) {
>> + error_propagate(errp, err);
>> + return;
>> + }
>> +
>> + if (!dev->has_timer_period) {
>> + dev->has_timer_period = true;
>> + dev->timer_period = 10000; /* 100Hz -> 10ms */
>> + }
>> +}
>
> Drive-by question: this validates just the generic part of Audiodev gets
> validated, not the driver-specific part. Is there nothing to validate?
> Does it happen somewhere else?
Driver specific validation/initialization happens in each driver's init
function (they're added in the upcoming patches).
>> +
>> +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);
>> + }
>> +}
>
> Your use of QAPI here looks good to me.
>
> [...]
>
^ permalink raw reply [flat|nested] 23+ messages in thread
* [Qemu-devel] [PATCH v4 05/14] alsaaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (3 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 04/14] audio: -audiodev command line option basic implementation Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 06/14] coreaudio: " Kővágó, Zoltán
` (9 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 344 ++++++++++++++-----------------------------
audio/audio_legacy.c | 94 +++++++++++-
2 files changed, 205 insertions(+), 233 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 8302f3e882..68fcfdd038 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 +387,19 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt,
- snd_pcm_format_t obtfmt)
+ snd_pcm_format_t obtfmt,
+ AudiodevPerDirectionOptions *pdo)
{
- dolog ("parameter | requested value | obtained value\n");
- dolog ("format | %10d | %10d\n", req->fmt, obtfmt);
- 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 " buffer count %" PRId32 "\n",
+ pdo->has_buffer_len ? pdo->buffer_len : 0,
+ pdo->has_buffer_count ? pdo->buffer_count : 0);
+ dolog("obtained: samples %ld\n", obt->samples);
}
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
@@ -451,23 +432,25 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
}
}
-static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep,
- ALSAConf *conf)
+static int alsa_open(bool in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ Audiodev *dev)
{
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
+ AudiodevAlsaOptions *aopts = &dev->u.alsa;
+ AudiodevAlsaPerDirectionOptions *apdo =
+ in ? aopts->alsa_in : aopts->alsa_out;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
- int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+ const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
nchannels = req->nchannels;
- size_in_usec = req->size_in_usec;
snd_pcm_hw_params_alloca (&hw_params);
@@ -527,79 +510,49 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (req->buffer_size) {
- unsigned long obt;
+ if (pdo->buffer_count) {
+ if (pdo->buffer_len) {
+ int64_t req = pdo->buffer_len * pdo->buffer_count;
- if (size_in_usec) {
int dir = 0;
- unsigned int btime = req->buffer_size;
+ unsigned int btime = req;
- err = snd_pcm_hw_params_set_buffer_time_near (
- handle,
- hw_params,
- &btime,
- &dir
- );
- obt = btime;
- }
- else {
- snd_pcm_uframes_t bsize = req->buffer_size;
+ err = snd_pcm_hw_params_set_buffer_time_near(
+ handle, hw_params, &btime, &dir);
- err = snd_pcm_hw_params_set_buffer_size_near (
- handle,
- hw_params,
- &bsize
- );
- obt = bsize;
- }
- if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
- size_in_usec ? "time" : "size", req->buffer_size);
- goto err;
- }
+ if (err < 0) {
+ alsa_logerr2(err, typ,
+ "Failed to set buffer time to %" PRId64 "\n",
+ req);
+ goto err;
+ }
- if ((req->override_mask & 2) && (obt - req->buffer_size))
- dolog ("Requested buffer %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->buffer_size, obt);
+ if (pdo->has_buffer_count && btime != req) {
+ dolog("Requested buffer time %" PRId64
+ " was rejected, using %u\n", req, btime);
+ }
+ } else {
+ dolog("Can't set buffer-count without buffer-len!\n");
+ }
}
- if (req->period_size) {
- unsigned long obt;
+ if (pdo->buffer_len) {
+ int dir = 0;
+ unsigned int ptime = pdo->buffer_len;
- if (size_in_usec) {
- int dir = 0;
- unsigned int ptime = req->period_size;
-
- err = snd_pcm_hw_params_set_period_time_near (
- handle,
- hw_params,
- &ptime,
- &dir
- );
- obt = ptime;
- }
- else {
- int dir = 0;
- snd_pcm_uframes_t psize = req->period_size;
-
- err = snd_pcm_hw_params_set_period_size_near (
- handle,
- hw_params,
- &psize,
- &dir
- );
- obt = psize;
- }
+ err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
+ &dir);
if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
- size_in_usec ? "time" : "size", req->period_size);
+ alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
+ pdo->buffer_len);
goto err;
}
- if (((req->override_mask & 1) && (obt - req->period_size)))
- dolog ("Requested period %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->period_size, obt);
+ if (pdo->has_buffer_len && ptime != pdo->buffer_len) {
+ dolog("Requested period time %" PRId32 " was rejected, using %d\n",
+ pdo->buffer_len, ptime);
+ }
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -631,33 +584,10 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf->threshold) {
- snd_pcm_uframes_t threshold;
- int bytes_per_sec;
-
- bytes_per_sec = freq << (nchannels == 2);
-
- switch (obt->fmt) {
- case AUDIO_FORMAT_S8:
- case AUDIO_FORMAT_U8:
- break;
-
- case AUDIO_FORMAT_S16:
- case AUDIO_FORMAT_U16:
- bytes_per_sec <<= 1;
- break;
-
- case AUDIO_FORMAT_S32:
- case AUDIO_FORMAT_U32:
- bytes_per_sec <<= 2;
- break;
-
- default:
- abort();
- }
-
- threshold = (conf->threshold * bytes_per_sec) / 1000;
- alsa_set_threshold (handle, threshold);
+ if (!in && aopts->has_threshold && aopts->threshold) {
+ struct audsettings as = { .freq = freq };
+ alsa_set_threshold(handle,
+ audio_buffer_frames(pdo, &as, aopts->threshold));
}
obt->nchannels = nchannels;
@@ -670,11 +600,11 @@ static int alsa_open (int in, struct alsa_params_req *req,
obt->nchannels != req->nchannels ||
obt->freq != req->freq) {
dolog ("Audio parameters for %s\n", typ);
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info(req, obt, obtfmt, pdo);
}
#ifdef DEBUG
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info(req, obt, obtfmt, pdo);
#endif
return 0;
@@ -800,19 +730,13 @@ 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 +757,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -873,16 +797,12 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.alsa_out;
switch (cmd) {
case VOICE_ENABLE:
{
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
+ bool poll_mode = apdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_out (hw)) {
@@ -911,19 +831,13 @@ 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 +858,7 @@ 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 +1000,12 @@ static int alsa_read (SWVoiceIn *sw, void *buf, int size)
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.alsa_in;
switch (cmd) {
case VOICE_ENABLE:
{
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
+ bool poll_mode = apdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_in (hw)) {
@@ -1118,88 +1028,59 @@ 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,
+ bool *has_apdo)
+{
+ if (!*has_apdo) {
+ *apdo = g_malloc0(sizeof(AudiodevAlsaPerDirectionOptions));
+ *has_apdo = true;
+ }
+
+ 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->alsa_in, &aopts->has_alsa_in);
+ alsa_init_per_direction(&aopts->alsa_out, &aopts->has_alsa_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->out->has_buffer_count) {
+ dev->out->buffer_count = 4;
+ }
+ if (!dev->out->has_buffer_len) {
+ /* 1024 frames assuming 44100Hz */
+ dev->out->buffer_len = 1024 * 1000000 / 44100;
+ }
+
+ /*
+ * OptsVisitor sets unspecified optional fields to zero, but do not depend
+ * on it...
+ */
+ if (!dev->in->has_buffer_count) {
+ dev->in->buffer_count = 0;
+ }
+ if (!dev->in->has_buffer_len) {
+ dev->in->buffer_len = 0;
+ }
+
+ return dev;
}
static void alsa_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option alsa_options[] = {
- {
- .name = "DAC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_out,
- .descr = "DAC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "DAC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_out,
- .descr = "DAC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_out_overridden
- },
- {
- .name = "DAC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_out,
- .descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_out_overridden
- },
- {
- .name = "ADC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_in,
- .descr =
- "ADC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "ADC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_in,
- .descr = "ADC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_in_overridden
- },
- {
- .name = "ADC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_in,
- .descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_in_overridden
- },
- {
- .name = "THRESHOLD",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.threshold,
- .descr = "(undocumented)"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_out,
- .descr = "DAC device name (for instance dmix)"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_in,
- .descr = "ADC device name"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
@@ -1217,7 +1098,6 @@ 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 c0b2c4604e..8319dc5240 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 +92,79 @@ 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(
+ AudiodevPerDirectionOptions *pdo, AudiodevAlsaPerDirectionOptions **apdo,
+ bool *has_apdo, const char *prefix)
+{
+ char buf[64];
+ size_t len = strlen(prefix);
+ bool size_in_usecs = false;
+ uint32_t buffer_size;
+ bool has_buffer_size = false;
+ bool dummy;
+
+ *apdo = g_malloc0(sizeof(AudiodevAlsaPerDirectionOptions));
+ *has_apdo = true;
+
+ 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, &pdo->buffer_len, &pdo->has_buffer_len);
+
+ if (pdo->has_buffer_len && !size_in_usecs) {
+ pdo->buffer_len = frames_to_usecs(pdo->buffer_len, pdo);
+ }
+
+ strcpy(buf + len, "BUFFER_SIZE");
+ get_int(buf, &buffer_size, &has_buffer_size);
+ if (has_buffer_size) {
+ if (!size_in_usecs) {
+ buffer_size = frames_to_usecs(buffer_size, pdo);
+ }
+
+ pdo->buffer_count = buffer_size;
+ pdo->has_buffer_count = true;
+ }
+}
+
+static void handle_alsa(Audiodev *dev)
+{
+ AudiodevAlsaOptions *aopt = &dev->u.alsa;
+ handle_alsa_per_direction(dev->in, &aopt->alsa_in, &aopt->has_alsa_in,
+ "QEMU_ALSA_ADC_");
+ handle_alsa_per_direction(dev->out, &aopt->alsa_out, &aopt->has_alsa_out,
+ "QEMU_ALSA_DAC_");
+
+ get_millis_to_usecs("QEMU_ALSA_THRESHOLD",
+ &aopt->threshold, &aopt->has_threshold);
+}
/* general */
static void handle_per_direction(
@@ -125,6 +208,15 @@ 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 06/14] coreaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (4 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 05/14] alsaaudio: port to -audiodev config Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 07/14] dsoundaudio: " Kővágó, Zoltán
` (8 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 24 ++++++++++++++++++++++
audio/coreaudio.c | 48 +++++++++-----------------------------------
2 files changed, 34 insertions(+), 38 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 8319dc5240..301934bf0c 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(
@@ -166,6 +177,15 @@ 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->out->buffer_len,
+ &dev->out->has_buffer_len, dev->out);
+ get_int("QEMU_COREAUDIO_BUFFER_COUNT",
+ &dev->out->buffer_count, &dev->out->has_buffer_count);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -213,6 +233,10 @@ 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..a7c1e418b6 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -36,11 +36,6 @@
#define MAC_OS_X_VERSION_10_6 1060
#endif
-typedef struct {
- int buffer_frames;
- int nbuffers;
-} CoreaudioConf;
-
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
@@ -507,7 +502,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
int err;
const char *typ = "playback";
AudioValueRange frameRange;
- CoreaudioConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
+ AudiodevPerDirectionOptions *pdo = dev->out;
+ int frames;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
@@ -538,16 +535,16 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
- if (frameRange.mMinimum > conf->buffer_frames) {
+ frames = audio_buffer_frames(pdo, as, 11610);
+ if (frameRange.mMinimum > frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
- }
- else if (frameRange.mMaximum < conf->buffer_frames) {
+ } else if (frameRange.mMaximum < frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
}
else {
- core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
+ core->audioDevicePropertyBufferFrameSize = frames;
}
/* set Buffer Frame Size */
@@ -568,7 +565,8 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
- hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
+ hw->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
+ core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
status = coreaudio_get_streamformat(core->outputDeviceID,
@@ -680,40 +678,15 @@ 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 +698,6 @@ 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 07/14] dsoundaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (5 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 06/14] coreaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 08/14] noaudio: " Kővágó, Zoltán
` (7 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 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 | 39 +++++++++++++++++++++++++++
audio/dsoundaudio.c | 59 ++++++++++++-----------------------------
3 files changed, 59 insertions(+), 45 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index b439f33f58..96181efb36 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -167,17 +167,18 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
dsound *s = drv_opaque;
WAVEFORMATEX wfx;
struct audsettings obt_as;
- DSoundConf *conf = &s->conf;
#ifdef DSBTYPE_IN
const char *typ = "ADC";
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
DSCBUFFERDESC bd;
DSCBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->in;
#else
const char *typ = "DAC";
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
DSBUFFERDESC bd;
DSBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->out;
#endif
if (!s->FIELD2) {
@@ -193,8 +194,8 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
memset (&bd, 0, sizeof (bd));
bd.dwSize = sizeof (bd);
bd.lpwfxFormat = &wfx;
+ bd.dwBufferBytes = audio_buffer_bytes(pdo, as, 92880);
#ifdef DSBTYPE_IN
- bd.dwBufferBytes = conf->bufsize_in;
hr = IDirectSoundCapture_CreateCaptureBuffer (
s->dsound_capture,
&bd,
@@ -203,7 +204,6 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
#else
bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
- bd.dwBufferBytes = conf->bufsize_out;
hr = IDirectSound_CreateSoundBuffer (
s->dsound,
&bd,
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 301934bf0c..0002fce627 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(
@@ -186,6 +210,17 @@ static void handle_coreaudio(Audiodev *dev)
&dev->out->buffer_count, &dev->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->out->buffer_len,
+ &dev->out->has_buffer_len, dev->out);
+ get_bytes_to_usecs("QEMU_DSOUND_BUFSIZE_IN", &dev->in->buffer_len,
+ &dev->in->has_buffer_len, dev->in);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -237,6 +272,10 @@ 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 08/14] noaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (6 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 07/14] dsoundaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 09/14] ossaudio: " Kővágó, Zoltán
` (6 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 09/14] ossaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (7 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 08/14] noaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 10/14] paaudio: " Kővágó, Zoltán
` (5 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 33 +++++++++
audio/ossaudio.c | 167 +++++++++++++++----------------------------
2 files changed, 90 insertions(+), 110 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 0002fce627..3d336259bb 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -221,6 +221,35 @@ static void handle_dsound(Audiodev *dev)
&dev->in->has_buffer_len, dev->in);
}
+/* OSS */
+static void handle_oss_per_direction(
+ AudiodevPerDirectionOptions *pdo, AudiodevOssPerDirectionOptions **opdo,
+ bool *has_opdo, const char *try_poll_env, const char *dev_env)
+{
+ *opdo = g_malloc0(sizeof(AudiodevOssPerDirectionOptions));
+ *has_opdo = true;
+
+ 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",
+ &pdo->buffer_len, &pdo->has_buffer_len, pdo);
+ get_int("QEMU_OSS_NFRAGS", &pdo->buffer_count, &pdo->has_buffer_count);
+}
+
+static void handle_oss(Audiodev *dev)
+{
+ AudiodevOssOptions *oopt = &dev->u.oss;
+ handle_oss_per_direction(dev->in, &oopt->oss_in, &oopt->has_oss_in,
+ "QEMU_AUDIO_ADC_TRY_POLL", "QEMU_OSS_ADC_DEV");
+ handle_oss_per_direction(dev->out, &oopt->oss_out, &oopt->has_oss_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)
@@ -276,6 +305,10 @@ 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..72b9dc3b5f 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 +252,26 @@ static int oss_get_version (int fd, int *version, const char *typ)
}
#endif
-static int oss_open (int in, struct oss_params *req,
- struct oss_params *obt, int *pfd, OSSConf* conf)
+static int oss_open(int in, struct oss_params *req, audsettings *as,
+ struct oss_params *obt, int *pfd, Audiodev *dev)
{
+ AudiodevOssOptions *oopts = &dev->u.oss;
+ AudiodevOssPerDirectionOptions *opdo = in ? oopts->oss_in : oopts->oss_out;
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
int fd;
- int oflags = conf->exclusive ? O_EXCL : 0;
+ int oflags = (oopts->has_exclusive && oopts->exclusive) ? O_EXCL : 0;
audio_buf_info abinfo;
int fmt, freq, nchannels;
int setfragment = 1;
- const char *dspname = in ? conf->devpath_in : conf->devpath_out;
+ const char *dspname = opdo->has_dev ? opdo->dev : "/dev/dsp";
const char *typ = in ? "ADC" : "DAC";
+#ifdef USE_DSP_POLICY
+ int policy = oopts->has_dsp_policy ? oopts->dsp_policy : 5;
+#endif
/* Kludge needed to have working mmap on Linux */
- oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+ oflags |= (oopts->has_try_mmap && oopts->try_mmap) ?
+ O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
@@ -285,6 +282,8 @@ static int oss_open (int in, struct oss_params *req,
freq = req->freq;
nchannels = req->nchannels;
fmt = req->fmt;
+ req->nfrags = pdo->has_buffer_count ? pdo->buffer_count : 4;
+ req->fragsize = audio_buffer_bytes(pdo, as, 23220);
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
@@ -308,18 +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.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.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 +820,42 @@ 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,
+ bool *has_opdo)
+{
+ if (!*has_opdo) {
+ *opdo = g_malloc0(sizeof(AudiodevOssPerDirectionOptions));
+ *has_opdo = true;
+ }
+
+ 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->oss_in, &oopts->has_oss_in);
+ oss_init_per_direction(&oopts->oss_out, &oopts->has_oss_out);
+
+ if (access(oopts->oss_in->has_dev ? oopts->oss_in->dev : "/dev/dsp",
+ R_OK | W_OK) < 0 ||
+ access(oopts->oss_out->has_dev ? oopts->oss_out->dev : "/dev/dsp",
+ R_OK | W_OK) < 0) {
return NULL;
}
- return conf;
+ return dev;
}
static void oss_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option oss_options[] = {
- {
- .name = "FRAGSIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.fragsize,
- .descr = "Fragment size in bytes"
- },
- {
- .name = "NFRAGS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.nfrags,
- .descr = "Number of fragments"
- },
- {
- .name = "MMAP",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.try_mmap,
- .descr = "Try using memory mapped access"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.devpath_out,
- .descr = "Path to DAC device"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.devpath_in,
- .descr = "Path to ADC device"
- },
- {
- .name = "EXCLUSIVE",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.exclusive,
- .descr = "Open device in exclusive mode (vmix won't work)"
- },
-#ifdef USE_DSP_POLICY
- {
- .name = "POLICY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.policy,
- .descr = "Set the timing policy of the device, -1 to use fragment mode",
- },
-#endif
- { /* End of list */ }
-};
-
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
@@ -925,7 +873,6 @@ 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 10/14] paaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (8 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 09/14] ossaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 11/14] sdlaudio: " Kővágó, Zoltán
` (4 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 39 +++++++++++++++++++
audio/paaudio.c | 91 +++++++++++++++++++-------------------------
2 files changed, 79 insertions(+), 51 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 3d336259bb..d8cd5e27f6 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;
@@ -250,6 +260,31 @@ 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, bool *has_ppdo, const char *env)
+{
+ *ppdo = g_malloc0(sizeof(AudiodevPaPerDirectionOptions));
+ *has_ppdo = true;
+
+ get_str(env, &(*ppdo)->name, &(*ppdo)->has_name);
+}
+
+static void handle_pa(Audiodev *dev)
+{
+ handle_pa_per_direction(&dev->u.pa.sink, &dev->u.pa.has_sink,
+ "QEMU_PA_SINK");
+ handle_pa_per_direction(&dev->u.pa.source, &dev->u.pa.has_source,
+ "QEMU_PA_SOURCE");
+
+ get_samples_to_usecs("QEMU_PA_SAMPLES", &dev->in->buffer_len,
+ &dev->in->has_buffer_len, dev->in);
+ get_samples_to_usecs("QEMU_PA_SAMPLES", &dev->out->buffer_len,
+ &dev->out->has_buffer_len, dev->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)
@@ -309,6 +344,10 @@ 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..18db0ae26c 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -2,6 +2,7 @@
#include "qemu/osdep.h"
#include "qemu-common.h"
#include "audio.h"
+#include "qapi/opts-visitor.h"
#include <pulse/pulseaudio.h>
@@ -10,14 +11,7 @@
#include "audio_pt_int.h"
typedef struct {
- int samples;
- char *server;
- char *sink;
- char *source;
-} PAConf;
-
-typedef struct {
- PAConf conf;
+ Audiodev *dev;
pa_threaded_mainloop *mainloop;
pa_context *context;
} paaudio;
@@ -32,6 +26,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
+ int samples;
} PAVoiceOut;
typedef struct {
@@ -46,6 +41,7 @@ typedef struct {
const void *read_data;
size_t read_index, read_length;
paaudio *g;
+ int samples;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -227,7 +223,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN(pa->live, pa->g->conf.samples >> 5);
+ decr = to_mix = audio_MIN(pa->live, pa->samples >> 5);
rpos = pa->rpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -319,7 +315,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN(pa->dead, pa->g->conf.samples >> 5);
+ incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5);
wpos = pa->wpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -546,6 +542,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
paaudio *g = pa->g = drv_opaque;
+ AudiodevPaOptions *popts = &g->dev->u.pa;
+ AudiodevPaPerDirectionOptions *ppdo = popts->sink;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -566,7 +564,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
g,
"qemu",
PA_STREAM_PLAYBACK,
- g->conf.sink,
+ ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
@@ -578,7 +576,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(g->dev->out, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -612,6 +611,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
paaudio *g = pa->g = drv_opaque;
+ AudiodevPaOptions *popts = &g->dev->u.pa;
+ AudiodevPaPerDirectionOptions *ppdo = popts->source;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -623,7 +624,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
g,
"qemu",
PA_STREAM_RECORD,
- g->conf.source,
+ ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
@@ -635,7 +636,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(g->dev->in, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -808,13 +810,22 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
}
/* common */
-static PAConf glob_conf = {
- .samples = 4096,
-};
+static void qpa_init_per_direction(AudiodevPaPerDirectionOptions **ppdo,
+ bool *has_ppdo)
+{
+ if (!*has_ppdo) {
+ *ppdo = g_malloc0(sizeof(AudiodevPaPerDirectionOptions));
+ *has_ppdo = true;
+ }
+}
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 +840,15 @@ static void *qpa_audio_init(Audiodev *dev)
}
}
- paaudio *g = g_malloc(sizeof(paaudio));
- g->conf = glob_conf;
+ assert(dev->driver == AUDIODEV_DRIVER_PA);
+
+ qpa_init_per_direction(&dev->u.pa.sink, &dev->u.pa.has_sink);
+ qpa_init_per_direction(&dev->u.pa.source, &dev->u.pa.has_source);
+
+ g = g_malloc(sizeof(paaudio));
+ server = popts->has_server ? popts->server : NULL;
+
+ g->dev = dev;
g->mainloop = NULL;
g->context = NULL;
@@ -840,14 +858,14 @@ 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 +928,6 @@ 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 +945,6 @@ 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 11/14] sdlaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (9 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 10/14] paaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 12/14] spiceaudio: " Kővágó, Zoltán
` (3 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 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 d8cd5e27f6..214774b947 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -285,6 +285,14 @@ 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->out->buffer_len,
+ &dev->out->has_buffer_len, dev->out);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -348,6 +356,10 @@ 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 097841fde1..cf6ac19927 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -49,12 +49,6 @@ typedef struct SDLVoiceOut {
int decr;
} SDLVoiceOut;
-static struct {
- int nb_samples;
-} conf = {
- .nb_samples = 1024
-};
-
static struct SDLAudioState {
int exit;
#if USE_SEMAPHORE
@@ -63,6 +57,7 @@ static struct SDLAudioState {
#endif
int initialized;
bool driver_created;
+ Audiodev *dev;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
@@ -392,7 +387,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
req.freq = as->freq;
req.format = aud_to_sdlfmt (as->fmt);
req.channels = as->nchannels;
- req.samples = conf.nb_samples;
+ req.samples = audio_buffer_samples(s->dev->out, as, 11610);
req.callback = sdl_callback;
req.userdata = sdl;
@@ -467,6 +462,7 @@ static void *sdl_audio_init(Audiodev *dev)
#endif
s->driver_created = true;
+ s->dev = dev;
return s;
}
@@ -480,18 +476,9 @@ static void sdl_audio_fini (void *opaque)
#endif
SDL_QuitSubSystem (SDL_INIT_AUDIO);
s->driver_created = false;
+ s->dev = NULL;
}
-static struct audio_option sdl_options[] = {
- {
- .name = "SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.nb_samples,
- .descr = "Size of SDL buffer in samples"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
@@ -503,7 +490,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
static struct audio_driver sdl_audio_driver = {
.name = "sdl",
.descr = "SDL http://www.libsdl.org",
- .options = sdl_options,
.init = sdl_audio_init,
.fini = sdl_audio_fini,
.pcm_ops = &sdl_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 23+ messages in thread* [Qemu-devel] [PATCH v4 12/14] spiceaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (10 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 11/14] sdlaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 13/14] wavaudio: " Kővágó, Zoltán
` (2 subsequent siblings)
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 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] 23+ messages in thread* [Qemu-devel] [PATCH v4 13/14] wavaudio: port to -audiodev config
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (11 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 12/14] spiceaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 14/14] audio: -audiodev command line option: cleanup Kővágó, Zoltán
2019-01-31 17:56 ` [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) no-reply
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 15 ++++++++++++
audio/wavaudio.c | 58 +++++++-------------------------------------
2 files changed, 24 insertions(+), 49 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 214774b947..2c2447176c 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -293,6 +293,17 @@ static void handle_sdl(Audiodev *dev)
&dev->out->has_buffer_len, dev->out);
}
+/* wav */
+static void handle_wav(Audiodev *dev)
+{
+ get_int("QEMU_WAV_FREQUENCY",
+ &dev->out->frequency, &dev->out->has_frequency);
+ get_fmt("QEMU_WAV_FORMAT", &dev->out->format, &dev->out->has_format);
+ get_int("QEMU_WAV_DAC_FIXED_CHANNELS",
+ &dev->out->channels, &dev->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)
@@ -360,6 +371,10 @@ 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..214e30ccd9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -24,6 +24,7 @@
#include "qemu/osdep.h"
#include "qemu/host-utils.h"
#include "qemu/timer.h"
+#include "qapi/opts-visitor.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -37,11 +38,6 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-typedef struct {
- struct audsettings settings;
- const char *wav_path;
-} WAVConf;
-
static int wav_run_out (HWVoiceOut *hw, int live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
@@ -112,8 +108,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
- WAVConf *conf = drv_opaque;
- struct audsettings wav_as = conf->settings;
+ Audiodev *dev = drv_opaque;
+ AudiodevWavOptions *wopts = &dev->u.wav;
+ struct audsettings wav_as = audiodev_to_audsettings(dev->out);
+ const char *wav_path = wopts->has_path ? wopts->path : "qemu.wav";
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
@@ -154,10 +152,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
le_store (hdr + 32, 1 << (bits16 + stereo), 2);
- wav->f = fopen (conf->wav_path, "wb");
+ wav->f = fopen(wav_path, "wb");
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
- conf->wav_path, strerror (errno));
+ wav_path, strerror(errno));
g_free (wav->pcm_buf);
wav->pcm_buf = NULL;
return -1;
@@ -225,54 +223,17 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static WAVConf glob_conf = {
- .settings.freq = 44100,
- .settings.nchannels = 2,
- .settings.fmt = AUDIO_FORMAT_S16,
- .wav_path = "qemu.wav"
-};
-
static void *wav_audio_init(Audiodev *dev)
{
- WAVConf *conf = g_malloc(sizeof(WAVConf));
- *conf = glob_conf;
- return conf;
+ assert(dev->driver == AUDIODEV_DRIVER_WAV);
+ return dev;
}
static void wav_audio_fini (void *opaque)
{
ldebug ("wav_fini");
- g_free(opaque);
}
-static struct audio_option wav_options[] = {
- {
- .name = "FREQUENCY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.settings.freq,
- .descr = "Frequency"
- },
- {
- .name = "FORMAT",
- .tag = AUD_OPT_FMT,
- .valp = &glob_conf.settings.fmt,
- .descr = "Format"
- },
- {
- .name = "DAC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.settings.nchannels,
- .descr = "Number of channels (1 - mono, 2 - stereo)"
- },
- {
- .name = "PATH",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.wav_path,
- .descr = "Path to wave file"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
@@ -284,7 +245,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
static struct audio_driver wav_audio_driver = {
.name = "wav",
.descr = "WAV renderer http://wikipedia.org/wiki/WAV",
- .options = wav_options,
.init = wav_audio_init,
.fini = wav_audio_fini,
.pcm_ops = &wav_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 23+ messages in thread* [Qemu-devel] [PATCH v4 14/14] audio: -audiodev command line option: cleanup
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (12 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 13/14] wavaudio: " Kővágó, Zoltán
@ 2019-01-28 22:43 ` Kővágó, Zoltán
2019-01-31 17:56 ` [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) no-reply
14 siblings, 0 replies; 23+ messages in thread
From: Kővágó, Zoltán @ 2019-01-28 22:43 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 | 201 +---------------------------------------------
2 files changed, 4 insertions(+), 214 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index cee46c4809..51963ccff8 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 b37c245a8a..fd6e54998a 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) {
@@ -297,89 +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)
-{
- char *optname;
- const char qemu_prefix[] = "QEMU_";
- size_t preflen, optlen;
-
- if (audio_bug(__func__, !prefix)) {
- dolog ("prefix = NULL\n");
- return;
- }
-
- if (audio_bug(__func__, !opt)) {
- dolog ("opt = NULL\n");
- return;
- }
-
- preflen = strlen (prefix);
-
- for (; opt->name; opt++) {
- size_t len, i;
- int def;
-
- if (!opt->valp) {
- dolog ("Option value pointer for `%s' is not set\n",
- opt->name);
- continue;
- }
-
- len = strlen (opt->name);
- /* len of opt->name + len of prefix + size of qemu_prefix
- * (includes trailing zero) + zero + underscore (on behalf of
- * sizeof) */
- optlen = len + preflen + sizeof (qemu_prefix) + 1;
- optname = g_malloc (optlen);
-
- pstrcpy (optname, optlen, qemu_prefix);
-
- /* copy while upper-casing, including trailing zero */
- for (i = 0; i <= preflen; ++i) {
- optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
- }
- pstrcat (optname, optlen, "_");
- pstrcat (optname, optlen, opt->name);
-
- def = 1;
- switch (opt->tag) {
- case AUD_OPT_BOOL:
- case AUD_OPT_INT:
- {
- int *intp = opt->valp;
- *intp = audio_get_conf_int (optname, *intp, &def);
- }
- break;
-
- case AUD_OPT_FMT:
- {
- AudioFormat *fmtp = opt->valp;
- *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
- }
- break;
-
- case AUD_OPT_STR:
- {
- const char **strp = opt->valp;
- *strp = audio_get_conf_str (optname, *strp, &def);
- }
- break;
-
- default:
- dolog ("Bad value tag for option `%s' - %d\n",
- optname, opt->tag);
- break;
- }
-
- if (!opt->overriddenp) {
- opt->overriddenp = &opt->overridden;
- }
- *opt->overriddenp = !def;
- g_free (optname);
- }
-}
-
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
@@ -1092,7 +902,7 @@ 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);
}
}
@@ -1137,7 +947,7 @@ 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);
}
}
@@ -1461,9 +1271,6 @@ 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) {
@@ -1490,11 +1297,11 @@ 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] 23+ messages in thread* Re: [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches)
2019-01-28 22:43 [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches) Kővágó, Zoltán
` (13 preceding siblings ...)
2019-01-28 22:43 ` [Qemu-devel] [PATCH v4 14/14] audio: -audiodev command line option: cleanup Kővágó, Zoltán
@ 2019-01-31 17:56 ` no-reply
14 siblings, 0 replies; 23+ messages in thread
From: no-reply @ 2019-01-31 17:56 UTC (permalink / raw)
To: dirty.ice.hu; +Cc: fam, qemu-devel, kraxel
Patchew URL: https://patchew.org/QEMU/cover.1548714862.git.DirtY.iCE.hu@gmail.com/
Hi,
This series seems to have some coding style problems. See output below for
more information:
Subject: [Qemu-devel] [PATCH v4 00/14] Audio patches (was: Audio 5.1 patches)
Type: series
Message-id: cover.1548714862.git.DirtY.iCE.hu@gmail.com
=== TEST SCRIPT BEGIN ===
#!/bin/bash
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
Switched to a new branch 'test'
9505d0ef85 audio: -audiodev command line option: cleanup
2f7a6267e8 wavaudio: port to -audiodev config
53f2b38246 spiceaudio: port to -audiodev config
83031a0586 sdlaudio: port to -audiodev config
a4cf2daf95 paaudio: port to -audiodev config
eca548aefa ossaudio: port to -audiodev config
96b7a11f8e noaudio: port to -audiodev config
01d330b3a7 dsoundaudio: port to -audiodev config
3738935850 coreaudio: port to -audiodev config
a2c88d6cd0 alsaaudio: port to -audiodev config
54e5804b4c audio: -audiodev command line option basic implementation
172a851936 audio: -audiodev command line option: documentation
a418745d8a audio: use qapi AudioFormat instead of audfmt_e
820a94e64b qapi: qapi for audio backends
=== OUTPUT BEGIN ===
1/14 Checking commit 820a94e64b20 (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, 246 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 a418745d8ae2 (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:140:
+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:2431:
+ case 0: vs->as.fmt = AUDIO_FORMAT_U8; break;
ERROR: trailing statements should be on next line
#1198: FILE: ui/vnc.c:2432:
+ case 1: vs->as.fmt = AUDIO_FORMAT_S8; break;
ERROR: trailing statements should be on next line
#1199: FILE: ui/vnc.c:2433:
+ case 2: vs->as.fmt = AUDIO_FORMAT_U16; break;
ERROR: trailing statements should be on next line
#1200: FILE: ui/vnc.c:2434:
+ case 3: vs->as.fmt = AUDIO_FORMAT_S16; break;
ERROR: trailing statements should be on next line
#1201: FILE: ui/vnc.c:2435:
+ case 4: vs->as.fmt = AUDIO_FORMAT_U32; break;
ERROR: trailing statements should be on next line
#1202: FILE: ui/vnc.c:2436:
+ 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 172a8519365e (audio: -audiodev command line option: documentation)
4/14 Checking commit 54e5804b4c2a (audio: -audiodev command line option basic implementation)
WARNING: added, moved or deleted file(s), does MAINTAINERS need updating?
#937:
new file mode 100644
total: 0 errors, 1 warnings, 1283 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 a2c88d6cd00e (alsaaudio: port to -audiodev config)
6/14 Checking commit 373893585072 (coreaudio: port to -audiodev config)
7/14 Checking commit 01d330b3a7f0 (dsoundaudio: port to -audiodev config)
8/14 Checking commit 96b7a11f8ee6 (noaudio: port to -audiodev config)
9/14 Checking commit eca548aefae2 (ossaudio: port to -audiodev config)
10/14 Checking commit a4cf2daf954f (paaudio: port to -audiodev config)
11/14 Checking commit 83031a0586cb (sdlaudio: port to -audiodev config)
12/14 Checking commit 53f2b3824652 (spiceaudio: port to -audiodev config)
13/14 Checking commit 2f7a6267e87e (wavaudio: port to -audiodev config)
14/14 Checking commit 9505d0ef8530 (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.1548714862.git.DirtY.iCE.hu@gmail.com/testing.checkpatch/?type=message.
---
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