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https://github.com/xemu-project/xemu.git
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b2f1c13c31
In commit26ec4e53f2
and similar commits we fixed the indentation for doc comments in our qapi json files to follow a new stricter standard for indentation, which permits only: @arg: description line 1 description line 2 or: @arg: line 1 line 2 Unfortunately since we didn't manage to get the script changes that enforced the new style in, a variety of commits (egdf4097aeaf
,2e44570321
) introduced new doc text which doesn't follow the new stricter rules for indentation on multi-line doc comments. Bring those into line with the new rules. Signed-off-by: Peter Maydell <peter.maydell@linaro.org> Message-Id: <20200810195019.25427-3-peter.maydell@linaro.org> Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Signed-off-by: Markus Armbruster <armbru@redhat.com>
387 lines
9.5 KiB
Python
387 lines
9.5 KiB
Python
# -*- mode: python -*-
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#
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# Copyright (C) 2015-2019 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
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#
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# This work is licensed under the terms of the GNU GPL, version 2 or later.
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# See the COPYING file in the top-level directory.
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##
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# @AudiodevPerDirectionOptions:
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#
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# General audio backend options that are used for both playback and
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# recording.
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#
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# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU and
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# convert audio formats when not supported by the backend. When
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# set to off, fixed-settings must be also off (default on,
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# since 4.2)
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#
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# @fixed-settings: use fixed settings for host input/output. When off,
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# frequency, channels and format must not be
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# specified (default true)
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#
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# @frequency: frequency to use when using fixed settings
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# (default 44100)
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#
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# @channels: number of channels when using fixed settings (default 2)
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#
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# @voices: number of voices to use (default 1)
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#
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# @format: sample format to use when using fixed settings
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# (default s16)
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#
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# @buffer-length: the buffer length in microseconds
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevPerDirectionOptions',
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'data': {
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'*mixing-engine': 'bool',
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'*fixed-settings': 'bool',
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'*frequency': 'uint32',
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'*channels': 'uint32',
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'*voices': 'uint32',
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'*format': 'AudioFormat',
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'*buffer-length': 'uint32' } }
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##
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# @AudiodevGenericOptions:
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#
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# Generic driver-specific options.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevGenericOptions',
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'data': {
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'*in': 'AudiodevPerDirectionOptions',
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'*out': 'AudiodevPerDirectionOptions' } }
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##
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# @AudiodevAlsaPerDirectionOptions:
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#
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# Options of the ALSA backend that are used for both playback and
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# recording.
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#
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# @dev: the name of the ALSA device to use (default 'default')
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#
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# @period-length: the period length in microseconds
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#
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# @try-poll: attempt to use poll mode, falling back to non-polling
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# access on failure (default true)
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevAlsaPerDirectionOptions',
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'base': 'AudiodevPerDirectionOptions',
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'data': {
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'*dev': 'str',
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'*period-length': 'uint32',
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'*try-poll': 'bool' } }
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##
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# @AudiodevAlsaOptions:
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#
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# Options of the ALSA audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# @threshold: set the threshold (in microseconds) when playback starts
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevAlsaOptions',
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'data': {
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'*in': 'AudiodevAlsaPerDirectionOptions',
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'*out': 'AudiodevAlsaPerDirectionOptions',
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'*threshold': 'uint32' } }
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##
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# @AudiodevCoreaudioPerDirectionOptions:
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#
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# Options of the Core Audio backend that are used for both playback and
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# recording.
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#
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# @buffer-count: number of buffers
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevCoreaudioPerDirectionOptions',
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'base': 'AudiodevPerDirectionOptions',
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'data': {
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'*buffer-count': 'uint32' } }
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##
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# @AudiodevCoreaudioOptions:
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#
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# Options of the coreaudio audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevCoreaudioOptions',
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'data': {
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'*in': 'AudiodevCoreaudioPerDirectionOptions',
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'*out': 'AudiodevCoreaudioPerDirectionOptions' } }
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##
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# @AudiodevDsoundOptions:
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#
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# Options of the DirectSound audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# @latency: add extra latency to playback in microseconds
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# (default 10000)
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevDsoundOptions',
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'data': {
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'*in': 'AudiodevPerDirectionOptions',
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'*out': 'AudiodevPerDirectionOptions',
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'*latency': 'uint32' } }
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##
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# @AudiodevJackPerDirectionOptions:
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#
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# Options of the JACK backend that are used for both playback and
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# recording.
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#
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# @server-name: select from among several possible concurrent server instances
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# (default: environment variable $JACK_DEFAULT_SERVER if set, else "default")
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#
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# @client-name: the client name to use. The server will modify this name to
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# create a unique variant, if needed unless @exact-name is true (default: the
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# guest's name)
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#
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# @connect-ports: if set, a regular expression of JACK client port name(s) to
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# monitor for and automatically connect to
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#
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# @start-server: start a jack server process if one is not already present
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# (default: false)
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#
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# @exact-name: use the exact name requested otherwise JACK automatically
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# generates a unique one, if needed (default: false)
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#
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# Since: 5.1
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##
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{ 'struct': 'AudiodevJackPerDirectionOptions',
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'base': 'AudiodevPerDirectionOptions',
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'data': {
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'*server-name': 'str',
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'*client-name': 'str',
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'*connect-ports': 'str',
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'*start-server': 'bool',
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'*exact-name': 'bool' } }
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##
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# @AudiodevJackOptions:
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#
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# Options of the JACK audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# Since: 5.1
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##
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{ 'struct': 'AudiodevJackOptions',
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'data': {
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'*in': 'AudiodevJackPerDirectionOptions',
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'*out': 'AudiodevJackPerDirectionOptions' } }
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##
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# @AudiodevOssPerDirectionOptions:
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#
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# Options of the OSS backend that are used for both playback and
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# recording.
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#
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# @dev: file name of the OSS device (default '/dev/dsp')
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#
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# @buffer-count: number of buffers
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#
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# @try-poll: attempt to use poll mode, falling back to non-polling
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# access on failure (default true)
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevOssPerDirectionOptions',
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'base': 'AudiodevPerDirectionOptions',
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'data': {
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'*dev': 'str',
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'*buffer-count': 'uint32',
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'*try-poll': 'bool' } }
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##
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# @AudiodevOssOptions:
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#
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# Options of the OSS audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# @try-mmap: try using memory-mapped access, falling back to
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# non-memory-mapped access on failure (default true)
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#
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# @exclusive: open device in exclusive mode (vmix won't work)
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# (default false)
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#
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# @dsp-policy: set the timing policy of the device (between 0 and 10,
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# where smaller number means smaller latency but higher
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# CPU usage) or -1 to use fragment mode (option ignored
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# on some platforms) (default 5)
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevOssOptions',
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'data': {
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'*in': 'AudiodevOssPerDirectionOptions',
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'*out': 'AudiodevOssPerDirectionOptions',
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'*try-mmap': 'bool',
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'*exclusive': 'bool',
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'*dsp-policy': 'uint32' } }
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##
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# @AudiodevPaPerDirectionOptions:
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#
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# Options of the Pulseaudio backend that are used for both playback and
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# recording.
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#
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# @name: name of the sink/source to use
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#
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# @stream-name: name of the PulseAudio stream created by qemu. Can be
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# used to identify the stream in PulseAudio when you
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# create multiple PulseAudio devices or run multiple qemu
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# instances (default: audiodev's id, since 4.2)
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#
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# @latency: latency you want PulseAudio to achieve in microseconds
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# (default 15000)
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevPaPerDirectionOptions',
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'base': 'AudiodevPerDirectionOptions',
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'data': {
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'*name': 'str',
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'*stream-name': 'str',
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'*latency': 'uint32' } }
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##
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# @AudiodevPaOptions:
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#
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# Options of the PulseAudio audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# @server: PulseAudio server address (default: let PulseAudio choose)
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevPaOptions',
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'data': {
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'*in': 'AudiodevPaPerDirectionOptions',
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'*out': 'AudiodevPaPerDirectionOptions',
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'*server': 'str' } }
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##
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# @AudiodevWavOptions:
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#
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# Options of the wav audio backend.
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#
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# @in: options of the capture stream
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#
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# @out: options of the playback stream
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#
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# @path: name of the wav file to record (default 'qemu.wav')
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#
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# Since: 4.0
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##
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{ 'struct': 'AudiodevWavOptions',
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'data': {
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'*in': 'AudiodevPerDirectionOptions',
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'*out': 'AudiodevPerDirectionOptions',
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'*path': 'str' } }
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##
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# @AudioFormat:
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#
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# An enumeration of possible audio formats.
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#
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# @u8: unsigned 8 bit integer
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#
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# @s8: signed 8 bit integer
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#
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# @u16: unsigned 16 bit integer
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#
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# @s16: signed 16 bit integer
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#
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# @u32: unsigned 32 bit integer
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#
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# @s32: signed 32 bit integer
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#
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# @f32: single precision floating-point (since 5.0)
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#
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# Since: 4.0
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##
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{ 'enum': 'AudioFormat',
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'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32', 'f32' ] }
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##
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# @AudiodevDriver:
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#
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# An enumeration of possible audio backend drivers.
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#
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# @jack: JACK audio backend (since 5.1)
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#
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# Since: 4.0
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##
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{ 'enum': 'AudiodevDriver',
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'data': [ 'none', 'alsa', 'coreaudio', 'dsound', 'jack', 'oss', 'pa',
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'sdl', 'spice', 'wav' ] }
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##
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# @Audiodev:
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#
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# Options of an audio backend.
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#
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# @id: identifier of the backend
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#
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# @driver: the backend driver to use
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#
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# @timer-period: timer period (in microseconds, 0: use lowest possible)
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#
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# Since: 4.0
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##
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{ 'union': 'Audiodev',
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'base': {
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'id': 'str',
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'driver': 'AudiodevDriver',
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'*timer-period': 'uint32' },
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'discriminator': 'driver',
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'data': {
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'none': 'AudiodevGenericOptions',
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'alsa': 'AudiodevAlsaOptions',
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'coreaudio': 'AudiodevCoreaudioOptions',
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'dsound': 'AudiodevDsoundOptions',
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'jack': 'AudiodevJackOptions',
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'oss': 'AudiodevOssOptions',
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'pa': 'AudiodevPaOptions',
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'sdl': 'AudiodevGenericOptions',
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'spice': 'AudiodevGenericOptions',
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'wav': 'AudiodevWavOptions' } }
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