linux系统音量问题汇总

音频工具安装：apt-get install alsa-utils

aplay播放

aplay -h
Usage: aplay [OPTION]... [FILE]...

-h, --help              help
    --version           print current version
-l, --list-devices      list all soundcards and digital audio devices
-L, --list-pcms         list device names
-D, --device=NAME       select PCM by name
-q, --quiet             quiet mode
-t, --file-type TYPE    file type (voc, wav, raw or au)
-c, --channels=#        channels
-f, --format=FORMAT     sample format (case insensitive)
-r, --rate=#            sample rate
-d, --duration=#        interrupt after # seconds
-s, --samples=#         interrupt after # samples per channel
-M, --mmap              mmap stream
-N, --nonblock          nonblocking mode
-F, --period-time=#     distance between interrupts is # microseconds
-B, --buffer-time=#     buffer duration is # microseconds
    --period-size=#     distance between interrupts is # frames
    --buffer-size=#     buffer duration is # frames
-A, --avail-min=#       min available space for wakeup is # microseconds
-R, --start-delay=#     delay for automatic PCM start is # microseconds 
                        (relative to buffer size if <= 0)
-T, --stop-delay=#      delay for automatic PCM stop is # microseconds from xrun
-v, --verbose           show PCM structure and setup (accumulative)
-V, --vumeter=TYPE      enable VU meter (TYPE: mono or stereo)
-I, --separate-channels one file for each channel
-i, --interactive       allow interactive operation from stdin
-m, --chmap=ch1,ch2,..  Give the channel map to override or follow
    --disable-resample  disable automatic rate resample
    --disable-channels  disable automatic channel conversions
    --disable-format    disable automatic format conversions
    --disable-softvol   disable software volume control (softvol)
    --test-position     test ring buffer position
    --test-coef=#       test coefficient for ring buffer position (default 8)
                        expression for validation is: coef * (buffer_size / 2)
    --test-nowait       do not wait for ring buffer - eats whole CPU
    --max-file-time=#   start another output file when the old file has recorded
                        for this many seconds
    --process-id-file   write the process ID here
    --use-strftime      apply the strftime facility to the output file name
    --dump-hw-params    dump hw_params of the device
    --fatal-errors      treat all errors as fatal
Recognized sample formats are: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE IEC958_SUBFRAME_LE IEC958_SUBFRAME_BE MU_LAW A_LAW IMA_ADPCM MPEG GSM S20_LE S20_BE U20_LE U20_BE SPECIAL S24_3LE S24_3BE U24_3LE U24_3BE S20_3LE S20_3BE U20_3LE U20_3BE S18_3LE S18_3BE U18_3LE U18_3BE G723_24 G723_24_1B G723_40 G723_40_1B DSD_U8 DSD_U16_LE DSD_U32_LE DSD_U16_BE DSD_U32_BE
Some of these may not be available on selected hardware
The available format shortcuts are:
-f cd (16 bit little endian, 44100, stereo)
-f cdr (16 bit big endian, 44100, stereo)
-f dat (16 bit little endian, 48000, stereo)

查看声卡：aplay -l

播放：aplay -D plughw:1,0 dome.wav

arecord录音

arecord -h
Usage: arecord [OPTION]... [FILE]...

-h, --help              help
    --version           print current version
-l, --list-devices      list all soundcards and digital audio devices
-L, --list-pcms         list device names
-D, --device=NAME       select PCM by name
-q, --quiet             quiet mode
-t, --file-type TYPE    file type (voc, wav, raw or au)
-c, --channels=#        channels
-f, --format=FORMAT     sample format (case insensitive)
-r, --rate=#            sample rate
-d, --duration=#        interrupt after # seconds
-s, --samples=#         interrupt after # samples per channel
-M, --mmap              mmap stream
-N, --nonblock          nonblocking mode
-F, --period-time=#     distance between interrupts is # microseconds
-B, --buffer-time=#     buffer duration is # microseconds
    --period-size=#     distance between interrupts is # frames
    --buffer-size=#     buffer duration is # frames
-A, --avail-min=#       min available space for wakeup is # microseconds
-R, --start-delay=#     delay for automatic PCM start is # microseconds 
                        (relative to buffer size if <= 0)
-T, --stop-delay=#      delay for automatic PCM stop is # microseconds from xrun
-v, --verbose           show PCM structure and setup (accumulative)
-V, --vumeter=TYPE      enable VU meter (TYPE: mono or stereo)
-I, --separate-channels one file for each channel
-i, --interactive       allow interactive operation from stdin
-m, --chmap=ch1,ch2,..  Give the channel map to override or follow
    --disable-resample  disable automatic rate resample
    --disable-channels  disable automatic channel conversions
    --disable-format    disable automatic format conversions
    --disable-softvol   disable software volume control (softvol)
    --test-position     test ring buffer position
    --test-coef=#       test coefficient for ring buffer position (default 8)
                        expression for validation is: coef * (buffer_size / 2)
    --test-nowait       do not wait for ring buffer - eats whole CPU
    --max-file-time=#   start another output file when the old file has recorded
                        for this many seconds
    --process-id-file   write the process ID here
    --use-strftime      apply the strftime facility to the output file name
    --dump-hw-params    dump hw_params of the device
    --fatal-errors      treat all errors as fatal
Recognized sample formats are: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE IEC958_SUBFRAME_LE IEC958_SUBFRAME_BE MU_LAW A_LAW IMA_ADPCM MPEG GSM S20_LE S20_BE U20_LE U20_BE SPECIAL S24_3LE S24_3BE U24_3LE U24_3BE S20_3LE S20_3BE U20_3LE U20_3BE S18_3LE S18_3BE U18_3LE U18_3BE G723_24 G723_24_1B G723_40 G723_40_1B DSD_U8 DSD_U16_LE DSD_U32_LE DSD_U16_BE DSD_U32_BE
Some of these may not be available on selected hardware
The available format shortcuts are:
-f cd (16 bit little endian, 44100, stereo)
-f cdr (16 bit big endian, 44100, stereo)
-f dat (16 bit little endian, 48000, stereo)

查看声卡：arecord -l

录音：arecord -D plughw:2,0 -r44100 -f S16_LE -c 2 demo.wav

parecord录音

parecord -h
parecord [options]
Capture audio data from a PulseAudio sound server and write it to a file.

  -h, --help                            Show this help
      --version                         Show version

  -r, --record                          Create a connection for recording
  -p, --playback                        Create a connection for playback

  -v, --verbose                         Enable verbose operations

  -s, --server=SERVER                   The name of the server to connect to
  -d, --device=DEVICE                   The name of the sink/source to connect to
  -n, --client-name=NAME                How to call this client on the server
      --stream-name=NAME                How to call this stream on the server
      --volume=VOLUME                   Specify the initial (linear) volume in range 0...65536
      --rate=SAMPLERATE                 The sample rate in Hz (defaults to 44100)
      --format=SAMPLEFORMAT             The sample type, one of s16le, s16be, u8, float32le,
                                        float32be, ulaw, alaw, s32le, s32be, s24le, s24be,
                                        s24-32le, s24-32be (defaults to s16ne)
      --channels=CHANNELS               The number of channels, 1 for mono, 2 for stereo
                                        (defaults to 2)
      --channel-map=CHANNELMAP          Channel map to use instead of the default
      --fix-format                      Take the sample format from the sink/source the stream is
                                        being connected to.
      --fix-rate                        Take the sampling rate from the sink/source the stream is
                                        being connected to.
      --fix-channels                    Take the number of channels and the channel map
                                        from the sink/source the stream is being connected to.
      --no-remix                        Don't upmix or downmix channels.
      --no-remap                        Map channels by index instead of name.
      --latency=BYTES                   Request the specified latency in bytes.
      --process-time=BYTES              Request the specified process time per request in bytes.
      --latency-msec=MSEC               Request the specified latency in msec.
      --process-time-msec=MSEC          Request the specified process time per request in msec.
      --property=PROPERTY=VALUE         Set the specified property to the specified value.
      --raw                             Record/play raw PCM data.
      --passthrough                     Passthrough data.
      --file-format[=FFORMAT]           Record/play formatted PCM data.
      --list-file-formats               List available file formats.
      --monitor-stream=INDEX            Record from the sink input with index INDEX.

播放录音：parecord test.wav

alsamixer音量调节

alsamixer 是Linux 音频架构 ALSA 中的 Alsa 工具的其中一个，用于配置音频的各个参数。alsamixer 是基于文本下的图形界面的，可以通过键盘的上下键，左右键等，很方便地设置需要的音量，开关某个 switch（开关）等等操作。

amixer音量调节

amixer -h
Usage: amixer <options> [command]

Available options:
  -h,--help       this help
  -c,--card N     select the card
  -D,--device N   select the device, default 'default'
  -d,--debug      debug mode
  -n,--nocheck    do not perform range checking
  -v,--version    print version of this program
  -q,--quiet      be quiet
  -i,--inactive   show also inactive controls
  -a,--abstract L select abstraction level (none or basic)
  -s,--stdin      Read and execute commands from stdin sequentially
  -R,--raw-volume Use the raw value (default)
  -M,--mapped-volume Use the mapped volume

Available commands:
  scontrols       show all mixer simple controls
  scontents       show contents of all mixer simple controls (default command)
  sset sID P      set contents for one mixer simple control
  sget sID        get contents for one mixer simple control
  controls        show all controls for given card
  contents        show contents of all controls for given card
  cset cID P      set control contents for one control
  cget cID        get control contents for one control

调节输出音量：amixer -c 2 cset numid=1,iface=MIXER,name='DAC Playback Volume' 240

pactl 音量调节

pactl -h           
pactl [options] stat
pactl [options] info
pactl [options] list [short] [TYPE]
pactl [options] exit
pactl [options] upload-sample FILENAME [NAME]
pactl [options] play-sample  NAME [SINK]
pactl [options] remove-sample  NAME
pactl [options] load-module  NAME [ARGS ...]
pactl [options] unload-module  NAME|#N
pactl [options] move-(sink-input|source-output) #N SINK|SOURCE
pactl [options] suspend-(sink|source) NAME|#N 1|0
pactl [options] set-card-profile  CARD PROFILE
pactl [options] set-default-(sink|source) NAME
pactl [options] set-(sink|source)-port NAME|#N PORT
pactl [options] set-(sink|source)-volume NAME|#N VOLUME [VOLUME ...]
pactl [options] set-(sink-input|source-output)-volume #N VOLUME [VOLUME ...]
pactl [options] set-(sink|source)-mute NAME|#N 1|0|toggle
pactl [options] set-(sink-input|source-output)-mute #N 1|0|toggle
pactl [options] set-sink-formats #N FORMATS
pactl [options] set-port-latency-offset CARD-NAME|CARD-#N PORT OFFSET
pactl [options] subscribe

The special names @DEFAULT_SINK@, @DEFAULT_SOURCE@ and @DEFAULT_MONITOR@
can be used to specify the default sink, source and monitor.

  -h, --help                            Show this help
      --version                         Show version

  -s, --server=SERVER                   The name of the server to connect to
  -n, --client-name=NAME                How to call this client on the server

设置绝对音量，0%-100%,1表示声卡号：pactl set-sink-volume 1 90%

设置相对音量，增大10%：pactl set-sink-volume 1 +10%

设置相对音量，减小10%：pactl set-sink-volume 1 -10%