「Advanced Linux Sound Architecture/設定例」の版間の差分
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| + | [[Category:サウンド]] |
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| − | #redirect[[en:Advanced Linux Sound Architecture/Example Configurations]] |
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| + | [[en:Advanced Linux Sound Architecture/Example Configurations]] |
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| + | [[it:Advanced Linux Sound Architecture/Example Configurations]] |
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| + | 以下は [[ALSA]] の高度な設定のガイドです。メインの記事にあるように設定は {{ic|/etc/asound.conf}} に記述します。以下の設定が機能するという保証はありません。 |
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| + | |||
| + | {{Note|Most things discussed here are much easier to accomplish using alsa plugins like upmix which are explained in the main article.}} |
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| + | |||
| + | ==dmix を使ってステレオ音源を 7.1 にアップミックスして 7.1 の音源はアップミックスしない== |
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| + | # 2008-11-15 |
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| + | # |
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| + | # This .asoundrc will allow the following: |
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| + | # |
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| + | # - upmix stereo files to 7.1 speakers. |
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| + | # - playback real 7.1 sounds, on 7.1 speakers, |
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| + | # - allow the playback of both stereo (upmixed) and surround(7.1) sources at the same time. |
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| + | # - use the 6th and 7th channel (side speakers) as a separate soundcard, i.e. for headphones |
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| + | # (This is called the "alternate" output throughout the file, device names prefixed with 'a') |
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| + | # - play mono sources in stereo (like skype & ekiga) on the alterate output |
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| + | # |
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| + | # Make sure you have "8 Channels" and NOT "6 Channels" selected in alsamixer! |
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| + | # |
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| + | # Please try the following commands, to make sure everything is working as it should. |
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| + | # |
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| + | # To test stereo upmix : speaker-test -c2 -Ddefault -twav |
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| + | # To test surround(5.1): speaker-test -c6 -Dplug:dmix6 -twav |
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| + | # To test surround(7.1): speaker-test -c6 -Dplug:dmix8 -twav |
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| + | # To test alternative output: speaker-test -c2 -Daduplex -twav |
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| + | # To test mono upmix: speaker-test -c1 -Dmonoduplex -twav |
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| + | # |
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| + | # |
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| + | # It may not work out of the box for all cards. If it doesnt work for you, read the comments throughout the file. |
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| + | # The basis of this file was written by wishie of #alsa, and then modified with info from various sources by |
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| + | # squisher. Svenstaro modified it for 7.1 output support. |
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| + | |||
| + | #Define the soundcard to use |
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| + | pcm.snd_card { |
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| + | type hw |
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| + | card 0 |
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| + | device 0 |
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| + | } |
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| + | |||
| + | # 8 channel dmix - output whatever audio, to all 8 speakers |
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| + | pcm.dmix8 { |
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| + | type dmix |
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| + | ipc_key 1024 |
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| + | ipc_key_add_uid false |
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| + | ipc_perm 0660 |
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| + | slave { |
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| + | pcm "snd_card" |
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| + | rate 48000 |
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| + | channels 8 |
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| + | period_time 0 |
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| + | period_size 1024 |
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| + | buffer_time 0 |
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| + | buffer_size 5120 |
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| + | } |
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| + | |||
| + | # Some cards, like the "nforce" variants require the following to be uncommented. |
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| + | # It routes the audio to the correct speakers. |
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| + | # bindings { |
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| + | # 0 0 |
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| + | # 1 1 |
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| + | # 2 4 |
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| + | # 3 5 |
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| + | # 4 2 |
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| + | # 5 3 |
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| + | # 6 6 |
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| + | # 7 7 |
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| + | # } |
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| + | } |
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| + | |||
| + | # upmixing - duplicate stereo data to all 8 channels |
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| + | pcm.ch71dup { |
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| + | type route |
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| + | slave.pcm dmix8 |
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| + | slave.channels 8 |
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| + | ttable.0.0 1 |
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| + | ttable.1.1 1 |
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| + | ttable.0.2 1 |
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| + | ttable.1.3 1 |
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| + | ttable.0.4 0.5 |
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| + | ttable.1.4 0.5 |
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| + | ttable.0.5 0.5 |
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| + | ttable.1.5 0.5 |
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| + | ttable.0.6 1 |
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| + | ttable.1.7 1 |
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| + | } |
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| + | |||
| + | # this creates a six channel soundcard |
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| + | # and outputs to the eight channel one |
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| + | # i.e. for usage in mplayer I had to define in ~/.mplayer/config: |
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| + | # ao=alsa:device=dmix6 |
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| + | # channels=6 |
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| + | pcm.dmix6 { |
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| + | type route |
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| + | slave.pcm dmix8 |
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| + | slave.channels 8 |
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| + | ttable.0.0 1 |
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| + | ttable.1.1 1 |
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| + | ttable.2.2 1 |
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| + | ttable.3.3 1 |
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| + | ttable.4.4 1 |
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| + | ttable.5.5 1 |
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| + | ttable.6.6 1 |
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| + | ttable.7.7 1 |
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| + | } |
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| + | |||
| + | # share the microphone, i.e. because virtualbox grabs it by default |
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| + | pcm.microphone { |
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| + | type dsnoop |
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| + | ipc_key 1027 |
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| + | slave { |
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| + | pcm "snd_card" |
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| + | } |
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| + | } |
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| + | |||
| + | # rate conversion, needed i.e. for wine |
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| + | pcm.2chplug { |
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| + | type plug |
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| + | slave.pcm "ch71dup" |
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| + | } |
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| + | pcm.a2chplug { |
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| + | type plug |
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| + | slave.pcm "dmix8" |
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| + | } |
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| + | |||
| + | # routes the channel for the alternative |
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| + | # 2 channel output, which becomes the 7th and 8th channel |
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| + | # on the real soundcard |
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| + | #pcm.alt2ch { |
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| + | # type route |
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| + | # slave.pcm "a2chplug" |
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| + | # slave.channels 8 |
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| + | # ttable.0.6 1 |
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| + | # ttable.1.7 1 |
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| + | #} |
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| + | |||
| + | # skype and ekiga are only mono, so route left channel to the right channel |
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| + | # note: this gets routed to the alternative 2 channels |
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| + | pcm.mono_playback { |
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| + | type route |
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| + | slave.pcm "a2chplug" |
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| + | slave.channels 8 |
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| + | # Send Skype channel 0 to the L and R speakers at full volume |
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| + | #ttable.0.6 1 |
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| + | #ttable.0.7 1 |
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| + | } |
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| + | |||
| + | # 'full-duplex' device for use with aoss |
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| + | pcm.duplex { |
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| + | type asym |
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| + | playback.pcm "2chplug" |
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| + | capture.pcm "microphone" |
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| + | } |
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| + | |||
| + | #pcm.aduplex { |
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| + | # type asym |
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| + | # playback.pcm "alt2ch" |
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| + | # capture.pcm "microphone" |
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| + | #} |
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| + | |||
| + | pcm.monoduplex { |
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| + | type asym |
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| + | playback.pcm "mono_playback" |
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| + | capture.pcm "microphone" |
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| + | } |
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| + | |||
| + | # for aoss |
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| + | pcm.dsp0 "duplex" |
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| + | ctl.mixer0 "duplex" |
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| + | |||
| + | # softvol manages volume in alsa |
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| + | # i.e. wine likes this |
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| + | pcm.mainvol { |
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| + | type softvol |
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| + | slave.pcm "duplex" |
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| + | control { |
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| + | name "2ch-Upmix Master" |
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| + | card 0 |
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| + | } |
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| + | } |
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| + | |||
| + | #pcm.!default "mainvol" |
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| + | |||
| + | # set the default device according to the environment |
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| + | # variable ALSA_DEFAULT_PCM and default to mainvol |
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| + | pcm.!default { |
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| + | @func refer |
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| + | name { @func concat |
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| + | strings [ "pcm." |
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| + | { @func getenv |
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| + | vars [ ALSA_DEFAULT_PCM ] |
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| + | default "mainvol" |
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| + | } |
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| + | ] |
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| + | } |
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| + | } |
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| + | |||
| + | # uncomment the following if you want to be able to control |
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| + | # the mixer device through environment variables as well |
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| + | #ctl.!default { |
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| + | # @func refer |
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| + | # name { @func concat |
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| + | # strings [ "ctl." |
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| + | # { @func getenv |
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| + | # vars [ ALSA_DEFAULT_CTL |
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| + | # ALSA_DEFAULT_PCM |
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| + | # ] |
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| + | # default "duplex" |
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| + | # } |
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| + | # ] |
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| + | # } |
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| + | #} |
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| + | |||
| + | ==Surround51 incl. ステレオのアップミックス & dmix, L/R を交換, 部屋のスピーカーの位置の調整== |
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| + | |||
| + | Bad practice but works fine for almost everything without additional per-program/file customization: |
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| + | pcm.!default { |
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| + | type route |
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| + | ## forwards to the mixer pcm defined below |
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| + | slave.pcm dmix51 |
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| + | slave.channels 6 |
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| + | |||
| + | ## "Native Channels" stereo, swap left/right |
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| + | ttable.0.1 1 |
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| + | ttable.1.0 1 |
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| + | ## original normal left/right commented out |
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| + | # ttable.0.0 1 |
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| + | # ttable.1.1 1 |
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| + | |||
| + | ## route "native surround" so it still works but weaken signal (+ RL/RF swap) |
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| + | ## because my rear speakers are more like random than really behind me |
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| + | ttable.2.3 0.7 |
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| + | ttable.3.2 0.7 |
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| + | ttable.4.4 0.7 |
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| + | ttable.5.5 0.7 |
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| + | |||
| + | ## stereo => quad speaker "upmix" for "rear" speakers + swap L/R |
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| + | ttable.0.3 1 |
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| + | ttable.1.2 1 |
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| + | |||
| + | ## stereo L+R => join to Center & Subwoofer 50%/50% |
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| + | ttable.0.4 0.5 |
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| + | ttable.1.4 0.5 |
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| + | ttable.0.5 0.5 |
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| + | ttable.1.5 0.5 |
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| + | ## to test: "$ speaker-test -c6 -twav" and: "$ speaker-test -c2 -twav" |
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| + | } |
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| + | |||
| + | pcm.dmix51 { |
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| + | type dmix |
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| + | ipc_key 1024 |
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| + | # let multiple users share |
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| + | ipc_key_add_uid false |
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| + | # IPC permissions (octal, default 0600) |
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| + | # I think changing this fixed something - but I'm not sure what. |
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| + | ipc_perm 0660 # |
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| + | slave { |
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| + | ## this is specific to my hda_intel. Often hd:0 is just allready it; To find: $ aplay -L |
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| + | pcm surround51 |
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| + | # this rate makes my soundcard crackle |
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| + | # rate 44100 |
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| + | # this rate stops flash in firefox from playing audio, but I do not need that |
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| + | rate 48000 |
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| + | channels 6 |
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| + | ## Any other values in the 4 lines below seem to make my soundcard crackle, too |
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| + | period_time 0 |
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| + | period_size 1024 |
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| + | buffer_time 0 |
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| + | buffer_size 4096 |
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| + | } |
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| + | } |
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2015年12月9日 (水) 01:08時点における版
以下は ALSA の高度な設定のガイドです。メインの記事にあるように設定は /etc/asound.conf に記述します。以下の設定が機能するという保証はありません。
dmix を使ってステレオ音源を 7.1 にアップミックスして 7.1 の音源はアップミックスしない
# 2008-11-15
#
# This .asoundrc will allow the following:
#
# - upmix stereo files to 7.1 speakers.
# - playback real 7.1 sounds, on 7.1 speakers,
# - allow the playback of both stereo (upmixed) and surround(7.1) sources at the same time.
# - use the 6th and 7th channel (side speakers) as a separate soundcard, i.e. for headphones
# (This is called the "alternate" output throughout the file, device names prefixed with 'a')
# - play mono sources in stereo (like skype & ekiga) on the alterate output
#
# Make sure you have "8 Channels" and NOT "6 Channels" selected in alsamixer!
#
# Please try the following commands, to make sure everything is working as it should.
#
# To test stereo upmix : speaker-test -c2 -Ddefault -twav
# To test surround(5.1): speaker-test -c6 -Dplug:dmix6 -twav
# To test surround(7.1): speaker-test -c6 -Dplug:dmix8 -twav
# To test alternative output: speaker-test -c2 -Daduplex -twav
# To test mono upmix: speaker-test -c1 -Dmonoduplex -twav
#
#
# It may not work out of the box for all cards. If it doesnt work for you, read the comments throughout the file.
# The basis of this file was written by wishie of #alsa, and then modified with info from various sources by
# squisher. Svenstaro modified it for 7.1 output support.
#Define the soundcard to use
pcm.snd_card {
type hw
card 0
device 0
}
# 8 channel dmix - output whatever audio, to all 8 speakers
pcm.dmix8 {
type dmix
ipc_key 1024
ipc_key_add_uid false
ipc_perm 0660
slave {
pcm "snd_card"
rate 48000
channels 8
period_time 0
period_size 1024
buffer_time 0
buffer_size 5120
}
# Some cards, like the "nforce" variants require the following to be uncommented.
# It routes the audio to the correct speakers.
# bindings {
# 0 0
# 1 1
# 2 4
# 3 5
# 4 2
# 5 3
# 6 6
# 7 7
# }
}
# upmixing - duplicate stereo data to all 8 channels
pcm.ch71dup {
type route
slave.pcm dmix8
slave.channels 8
ttable.0.0 1
ttable.1.1 1
ttable.0.2 1
ttable.1.3 1
ttable.0.4 0.5
ttable.1.4 0.5
ttable.0.5 0.5
ttable.1.5 0.5
ttable.0.6 1
ttable.1.7 1
}
# this creates a six channel soundcard
# and outputs to the eight channel one
# i.e. for usage in mplayer I had to define in ~/.mplayer/config:
# ao=alsa:device=dmix6
# channels=6
pcm.dmix6 {
type route
slave.pcm dmix8
slave.channels 8
ttable.0.0 1
ttable.1.1 1
ttable.2.2 1
ttable.3.3 1
ttable.4.4 1
ttable.5.5 1
ttable.6.6 1
ttable.7.7 1
}
# share the microphone, i.e. because virtualbox grabs it by default
pcm.microphone {
type dsnoop
ipc_key 1027
slave {
pcm "snd_card"
}
}
# rate conversion, needed i.e. for wine
pcm.2chplug {
type plug
slave.pcm "ch71dup"
}
pcm.a2chplug {
type plug
slave.pcm "dmix8"
}
# routes the channel for the alternative
# 2 channel output, which becomes the 7th and 8th channel
# on the real soundcard
#pcm.alt2ch {
# type route
# slave.pcm "a2chplug"
# slave.channels 8
# ttable.0.6 1
# ttable.1.7 1
#}
# skype and ekiga are only mono, so route left channel to the right channel
# note: this gets routed to the alternative 2 channels
pcm.mono_playback {
type route
slave.pcm "a2chplug"
slave.channels 8
# Send Skype channel 0 to the L and R speakers at full volume
#ttable.0.6 1
#ttable.0.7 1
}
# 'full-duplex' device for use with aoss
pcm.duplex {
type asym
playback.pcm "2chplug"
capture.pcm "microphone"
}
#pcm.aduplex {
# type asym
# playback.pcm "alt2ch"
# capture.pcm "microphone"
#}
pcm.monoduplex {
type asym
playback.pcm "mono_playback"
capture.pcm "microphone"
}
# for aoss
pcm.dsp0 "duplex"
ctl.mixer0 "duplex"
# softvol manages volume in alsa
# i.e. wine likes this
pcm.mainvol {
type softvol
slave.pcm "duplex"
control {
name "2ch-Upmix Master"
card 0
}
}
#pcm.!default "mainvol"
# set the default device according to the environment
# variable ALSA_DEFAULT_PCM and default to mainvol
pcm.!default {
@func refer
name { @func concat
strings [ "pcm."
{ @func getenv
vars [ ALSA_DEFAULT_PCM ]
default "mainvol"
}
]
}
}
# uncomment the following if you want to be able to control
# the mixer device through environment variables as well
#ctl.!default {
# @func refer
# name { @func concat
# strings [ "ctl."
# { @func getenv
# vars [ ALSA_DEFAULT_CTL
# ALSA_DEFAULT_PCM
# ]
# default "duplex"
# }
# ]
# }
#}
Surround51 incl. ステレオのアップミックス & dmix, L/R を交換, 部屋のスピーカーの位置の調整
Bad practice but works fine for almost everything without additional per-program/file customization:
pcm.!default {
type route
## forwards to the mixer pcm defined below
slave.pcm dmix51
slave.channels 6
## "Native Channels" stereo, swap left/right
ttable.0.1 1
ttable.1.0 1
## original normal left/right commented out
# ttable.0.0 1
# ttable.1.1 1
## route "native surround" so it still works but weaken signal (+ RL/RF swap)
## because my rear speakers are more like random than really behind me
ttable.2.3 0.7
ttable.3.2 0.7
ttable.4.4 0.7
ttable.5.5 0.7
## stereo => quad speaker "upmix" for "rear" speakers + swap L/R
ttable.0.3 1
ttable.1.2 1
## stereo L+R => join to Center & Subwoofer 50%/50%
ttable.0.4 0.5
ttable.1.4 0.5
ttable.0.5 0.5
ttable.1.5 0.5
## to test: "$ speaker-test -c6 -twav" and: "$ speaker-test -c2 -twav"
}
pcm.dmix51 {
type dmix
ipc_key 1024
# let multiple users share
ipc_key_add_uid false
# IPC permissions (octal, default 0600)
# I think changing this fixed something - but I'm not sure what.
ipc_perm 0660 #
slave {
## this is specific to my hda_intel. Often hd:0 is just allready it; To find: $ aplay -L
pcm surround51
# this rate makes my soundcard crackle
# rate 44100
# this rate stops flash in firefox from playing audio, but I do not need that
rate 48000
channels 6
## Any other values in the 4 lines below seem to make my soundcard crackle, too
period_time 0
period_size 1024
buffer_time 0
buffer_size 4096
}
}