# `alsaaudio`

The `alsaaudio` module defines functions and classes for using ALSA.

alsaaudio.pcms(pcmtype=PCM_PLAYBACK)

  List available PCM devices by name.

  Arguments are:

  - pcmtype - can be either `PCM_CAPTURE` or `PCM_PLAYBACK` (default).

  Note:

  For `PCM_PLAYBACK`, the list of device names should be equivalent to the list of device names that `aplay -L` displays on the commandline:

      $ aplay -L

  For `PCM_CAPTURE`, the list of device names should be equivalent to the list of device names that `arecord -L` displays on the commandline:

      $ arecord -L

  New in 0.8

alsaaudio.cards()

  List the available ALSA cards by name. This function is only moderately useful. If you want to see a list of available PCM devices, use `pcms()` instead.

alsaaudio.mixers(cardindex=-1, device='default')

  List the available mixers. The arguments are:

  - cardindex - the card index. If this argument is given, the device name is constructed as: ‘hw:cardindex’ and the device keyword argument is ignored. `0` is the first hardware sound card.

    Note: This should not be used, as it bypasses most of ALSA’s configuration.

  - device - the name of the device on which the mixer resides. The default is `'default'`.

  Note: For a list of available controls, you can also use `amixer` on the commandline:

      $ amixer

  To elaborate the example, calling `mixers()` with the argument `cardindex=0` should give the same list of Mixer controls as:

      $ amixer -c 0

  And calling `mixers()` with the argument `device='foo'` should give the same list of Mixer controls as:

      $ amixer -D foo

  Changed in 0.8:

  - The keyword argument device is new and can be used to select virtual devices. As a result, the default behaviour has subtly changed. Since 0.8, this functions returns the mixers for the default device, not the mixers for the first card.

alsaaudio.asoundlib_version()

  Return a Python string containing the ALSA version found.

## PCM Objects

PCM objects in `alsaaudio` can play or capture (record) PCM sound through speakers or a microphone. The PCM constructor takes the following arguments:

class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1)

  This class is used to represent a PCM device (either for playback and recording). The arguments are:

  - type - can be either `PCM_CAPTURE` or `PCM_PLAYBACK` (default).

  - mode - can be either `PCM_NONBLOCK`, or `PCM_NORMAL` (default).

  - rate - the sampling rate in Hz. Typical values are `8000` (mainly used for telephony), `16000`, `44100` (default), `48000` and `96000`.

  - channels - the number of channels. The default value is 2 (stereo).

  - format - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures. The default value is `PCM_FORMAT_S16_LE`.

  Format | Description
  --- | ---
  `PCM_FORMAT_S8` | Signed 8 bit samples for each channel
  `PCM_FORMAT_U8` | Unsigned 8 bit samples for each channel
  `PCM_FORMAT_S16_LE` | Signed 16 bit samples for each channel Little Endian byte order)
  `PCM_FORMAT_S16_BE` | Signed 16 bit samples for each channel (Big Endian byte order)
  `PCM_FORMAT_U16_LE` | Unsigned 16 bit samples for each channel (Little Endian byte order)
  `PCM_FORMAT_U16_BE` | Unsigned 16 bit samples for each channel (Big Endian byte order)
  `PCM_FORMAT_S24_LE` | Signed 24 bit samples for each channel (Little Endian byte order in 4 bytes)
  `PCM_FORMAT_S24_BE` | Signed 24 bit samples for each channel (Big Endian byte order in 4 bytes)
  `PCM_FORMAT_U24_LE` | Unsigned 24 bit samples for each channel (Little Endian byte order in 4 bytes)
  `PCM_FORMAT_U24_BE` | Unsigned 24 bit samples for each channel (Big Endian byte order in 4 bytes)
  `PCM_FORMAT_S32_LE` | Signed 32 bit samples for each channel (Little Endian byte order)
  `PCM_FORMAT_S32_BE` | Signed 32 bit samples for each channel (Big Endian byte order)
  `PCM_FORMAT_U32_LE` | Unsigned 32 bit samples for each channel (Little Endian byte order)
  `PCM_FORMAT_U32_BE` | Unsigned 32 bit samples for each channel (Big Endian byte order)
  `PCM_FORMAT_FLOAT_LE` | 32 bit samples encoded as float (Little Endian byte order)
  `PCM_FORMAT_FLOAT_BE` | 32 bit samples encoded as float (Big Endian byte order)
  `PCM_FORMAT_FLOAT64_LE` | 64 bit samples encoded as float (Little Endian byte order)
  `PCM_FORMAT_FLOAT64_BE` | 64 bit samples encoded as float (Big Endian byte order)
  `PCM_FORMAT_MU_LAW` | A logarithmic encoding (used by Sun .au files and telephony)
  `PCM_FORMAT_A_LAW` | Another logarithmic encoding
  `PCM_FORMAT_IMA_ADPCM` | A 4:1 compressed format defined by the Interactive Multimedia Association.
  `PCM_FORMAT_MPEG` | MPEG encoded audio?
  `PCM_FORMAT_GSM` | 9600 bits/s constant rate encoding for speech
  `PCM_FORMAT_S24_3LE` | Signed 24 bit samples for each channel (Little Endian byte order in 3 bytes)
  `PCM_FORMAT_S24_3BE` | Signed 24 bit samples for each channel (Big Endian byte order in 3 bytes)
  `PCM_FORMAT_U24_3LE` | Unsigned 24 bit samples for each channel (Little Endian byte order in 3 bytes)
  `PCM_FORMAT_U24_3BE` | Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)

  - periodsize - the period size in frames. Make sure you understand the meaning of periods. The default value is 32, which is below the actual minimum of most devices, and will therefore likely be larger in practice.

  - periods - the number of periods in the buffer. The default value is 4.

  - device - the name of the PCM device that should be used (for example a value from the output of `pcms()`). The default value is `'default'`.

  - cardindex - the card index. If this argument is given, the device name is constructed as ‘hw:cardindex’ and the device keyword argument is ignored. `0` is the first hardware sound card.

    Note: This should not be used, as it bypasses most of ALSA’s configuration.

  This will construct a PCM object with the given settings.

  Changed in 0.10:

  - Added the optional named parameter periods.

  Changed in 0.9:

  - Added the optional named parameters rate, channels, format and periodsize.

  Changed in 0.8:

  - The card keyword argument is still supported, but deprecated. Please use device instead.

  - The keyword argument cardindex was added.

  The card keyword is deprecated because it guesses the real ALSA name of the card. This was always fragile and broke some legitimate usecases.

PCM objects have the following methods:

PCM.info()

  The info function returns a dictionary containing the configuration of a PCM device. As ALSA takes into account limitations of the hardware and software devices the configuration achieved might not correspond to the values used during creation. There is therefore a need to check the realised configuration before processing the sound coming from the device or before sending sound to a device. A small subset of parameters can be set, but cannot be queried. These parameters are stored by alsaaudio and returned as they were given by the user, to distinguish them from parameters retrieved from ALSA these parameters have a name prefixed with “ (call value) “. Yet another set of properties derives directly from the hardware and can be obtained through ALSA.

  Key | Description (Reference) | Type
  --- | --- | ---
  name | PCM():device | string
  card_no | index of card | integer (negative indicates device not associable with a card)
  device_no | index of PCM device | integer
  subdevice_no | index of PCM subdevice | integer
  state | name of PCM state | string
  access_type | name of PCM access type | string
  (call value) type | PCM():type | integer
  (call value) type_name | PCM():type | string
  (call value) mode | PCM():mode | integer
  (call value) mode_name | PCM():mode | string
  format | PCM():format | integer
  format_name | PCM():format | string
  format_description | PCM():format | string
  subformat_name | name of PCM subformat | string
  subformat_description | description of subformat | string
  channels | PCM():channels | integer
  rate | PCM():rate | integer (Hz)
  period_time | period duration | integer (\(\mu s\))
  period_size | PCM():period_size | integer (frames)
  buffer_time | buffer time | integer (\(\mu s\)) (negative indicates error)
  buffer_size | buffer size | integer (frames) (negative indicates error)
  get_periods | approx. periods in buffer | integer (negative indicates error)
  rate_numden | numerator, denominator | tuple (integer (Hz), integer (Hz))
  significant_bits | significant bits in sample | integer (negative indicates error)
  is_batch | hw: double buffering | boolean (True: hardware supported)
  is_block_transfer | hw: block transfer | boolean (True: hardware supported)
  is_double | hw: double buffering | boolean (True: hardware supported)
  is_half_duplex | hw: half-duplex | boolean (True: hardware supported)
  is_joint_duplex | hw: joint-duplex | boolean (True: hardware supported)
  can_overrange | hw: overrange detection | boolean (True: hardware supported)
  can_mmap_sample_resolution | hw: sample-resol. mmap | boolean (True: hardware supported)
  can_pause | hw: pause | boolean (True: hardware supported)
  can_resume | hw: resume | boolean (True: hardware supported)
  can_sync_start | hw: synchronized start | boolean (True: hardware supported)

  The italicized descriptions give a summary of the “full” description as it can be found in the ALSA documentation. “hw:”: indicates that the property indicated relates to the hardware. Parameters passed to the PCM object during instantation are prefixed with “PCM():”, they are described there for the keyword argument indicated after “PCM():”.

PCM.pcmtype()

  Returns the type of PCM object. Either `PCM_CAPTURE` or `PCM_PLAYBACK`.

PCM.pcmmode()

  Return the mode of the PCM object. One of `PCM_NONBLOCK`, `PCM_ASYNC`, or `PCM_NORMAL`

PCM.cardname()

  Return the name of the sound card used by this PCM object.

PCM.setchannels(nchannels)

  Deprecated since version 0.9: Use the channels named argument to `PCM()`.

PCM.setrate(rate)

  Deprecated since version 0.9: Use the rate named argument to `PCM()`.

PCM.setformat(format)

  Deprecated since version 0.9: Use the format named argument to `PCM()`.

PCM.setperiodsize(period)

  Deprecated since version 0.9: Use the periodsize named argument to `PCM()`.

PCM.info()

  Returns a dictionary with the PCM object’s configured parameters.

  Values are retrieved from the ALSA library if they are available; otherwise they represent those stored by pyalsaaudio, and their keys are prefixed with ‘ (call value) ‘.

  New in 0.9.1

PCM.dumpinfo()

  Dumps the PCM object’s configured parameters to stdout.

PCM.state()

  Returs the current state of the stream, which can be one of `PCM_STATE_OPEN` (this should not actually happen), `PCM_STATE_SETUP` (after `drop()` or `drain()`), `PCM_STATE_PREPARED` (after construction), `PCM_STATE_RUNNING`, `PCM_STATE_XRUN`, `PCM_STATE_DRAINING`, `PCM_STATE_PAUSED`, `PCM_STATE_SUSPENDED`, and `PCM_STATE_DISCONNECTED`.

  New in 0.10

PCM.read()

  In `PCM_NORMAL` mode, this function blocks until a full period is available, and then returns a tuple (length,data) where length is the number of frames of captured data, and data is the captured sound frames as a string. The length of the returned data will be periodsize*framesize bytes.

  In `PCM_NONBLOCK` mode, the call will not block, but will return `(0,'')` if no new period has become available since the last call to read.

  In case of an overrun, this function will return a negative size: `-EPIPE`. This indicates that data was lost, even if the operation itself succeeded. Try using a larger periodsize.

PCM.write(data)

  Writes (plays) the sound in data. The length of data must be a multiple of the frame size, and should be exactly the size of a period. If less than ‘period size’ frames are provided, the actual playout will not happen until more data is written.

  If the device is not in `PCM_NONBLOCK` mode, this call will block if the kernel buffer is full, and until enough sound has been played to allow the sound data to be buffered. The call always returns the size of the data provided.

  In `PCM_NONBLOCK` mode, the call will return immediately, with a return value of zero, if the buffer is full. In this case, the data should be written at a later time.

  Note that this call completing means only that the samples were buffered in the kernel, and playout will continue afterwards. Make sure that the stream is drained before discarding the PCM handle.

PCM.pause([enable=True])

  If enable is `True`, playback or capture is paused. Otherwise, playback/capture is resumed.

PCM.drop()

  Stop the stream and drop residual buffered frames.

  New in 0.9

PCM.drain()

  For `PCM_PLAYBACK` PCM objects, play residual buffered frames and then stop the stream. In `PCM_NORMAL` mode, this function blocks until all pending playback is drained.

  For `PCM_CAPTURE` PCM objects, this function is not very useful.

  New in 0.10

PCM.close()

  Closes the PCM device.

  For `PCM_PLAYBACK` PCM objects in `PCM_NORMAL` mode, this function blocks until all pending playback is drained.

PCM.polldescriptors()

  Returns a list of tuples of (file descriptor, eventmask) that can be used to wait for changes on the PCM with select.poll.

  The eventmask value is compatible with `poll.register`__ in the Python `select` module.

PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE])

  Set the ALSA timestamp mode on the device. The mode argument can be set to either `PCM_TSTAMP_NONE` or `PCM_TSTAMP_ENABLE`.

PCM.get_tstamp_mode()

  Return the integer value corresponding to the ALSA timestamp mode. The return value can be either `PCM_TSTAMP_NONE` or `PCM_TSTAMP_ENABLE`.

PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY])

  Set the ALSA timestamp mode on the device. The type argument can be set to either `PCM_TSTAMP_TYPE_GETTIMEOFDAY`, `PCM_TSTAMP_TYPE_MONOTONIC` or `PCM_TSTAMP_TYPE_MONOTONIC_RAW`.

PCM.get_tstamp_type()

  Return the integer value corresponding to the ALSA timestamp type. The return value can be either `PCM_TSTAMP_TYPE_GETTIMEOFDAY`, `PCM_TSTAMP_TYPE_MONOTONIC` or `PCM_TSTAMP_TYPE_MONOTONIC_RAW`.

PCM.htimestamp()

  Return a Python tuple (seconds, nanoseconds, frames_available_in_buffer).

  The type of output is controlled by the tstamp_type, as described in the table below.

  Timestamp Type | Description
  --- | ---
  `PCM_TSTAMP_TYPE_GETTIMEOFDAY` | System-wide realtime clock with seconds since epoch.
  `PCM_TSTAMP_TYPE_MONOTONIC` | Monotonic time from an unspecified starting time. Progress is NTP synchronized.
  `PCM_TSTAMP_TYPE_MONOTONIC_RAW` | Monotonic time from an unspecified starting time using only the system clock.

  The timestamp mode is controlled by the tstamp_mode, as described in the table below.

  Timestamp Mode | Description
  --- | ---
  `PCM_TSTAMP_NONE` | No timestamp.
  `PCM_TSTAMP_ENABLE` | Update timestamp at every hardware position update.

A few hints on using PCM devices for playback

The most common reason for problems with playback of PCM audio is that writes to PCM devices must exactly match the data rate of the device.

If too little data is written to the device, it will underrun, and ugly clicking sounds will occur. Conversely, of too much data is written to the device, the write function will either block (`PCM_NORMAL` mode) or return zero (`PCM_NONBLOCK` mode).

If your program does nothing but play sound, the best strategy is to put the device in `PCM_NORMAL` mode, and just write as much data to the device as possible. This strategy can also be achieved by using a separate thread with the sole task of playing out sound.

In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few periods by calling write a couple of times, and then use some timer method to write one period size of data to the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer doesn’t expire exactly on time.

Also note, that most timer APIs that you can find for Python will accummulate time delays: If you set the timer to expire after 1/10’th of a second, the actual timeout will happen slightly later, which will accumulate to quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add extra writes as nessecary.

## Mixer Objects

Mixer objects provides access to the ALSA mixer API.

class alsaaudio.Mixer(control='Master', id=0, cardindex=-1, device='default')

  Arguments are:

  - control - specifies which control to manipulate using this mixer object. The list of available controls can be found with the `alsaaudio`.`mixers()` function. The default value is `'Master'` - other common controls may be `'Master Mono'`, `'PCM'`, `'Line'`, etc.

  - id - the id of the mixer control. Default is `0`.

  - cardindex - specifies which card should be used. If this argument is given, the device name is constructed like this: ‘hw:cardindex’ and the device keyword argument is ignored. `0` is the first sound card.

  - device - the name of the device on which the mixer resides. The default value is `'default'`.

  Changed in 0.8:

  - The keyword argument device is new and can be used to select virtual devices.

Mixer objects have the following methods:

Mixer.cardname()

  Return the name of the sound card used by this Mixer object

Mixer.mixer()

  Return the name of the specific mixer controlled by this object, For example `'Master'` or `'PCM'`

Mixer.mixerid()

  Return the ID of the ALSA mixer controlled by this object.

Mixer.switchcap()

  Returns a list of the switches which are defined by this specific mixer. Possible values in this list are:

  Switch | Description
  --- | ---
  ‘Mute’ | This mixer can mute
  ‘Joined Mute’ | This mixer can mute all channels at the same time
  ‘Playback Mute’ | This mixer can mute the playback output
  ‘Joined Playback Mute’ | Mute playback for all channels at the same time}
  ‘Capture Mute’ | Mute sound capture
  ‘Joined Capture Mute’ | Mute sound capture for all channels at a time}
  ‘Capture Exclusive’ | Not quite sure what this is

  To manipulate these switches use the `setrec()` or `setmute()` methods

Mixer.volumecap()

  Returns a list of the volume control capabilities of this mixer. Possible values in the list are:

  Capability | Description
  --- | ---
  ‘Volume’ | This mixer can control volume
  ‘Joined Volume’ | This mixer can control volume for all channels at the same time
  ‘Playback Volume’ | This mixer can manipulate the playback output
  ‘Joined Playback Volume’ | Manipulate playback volumne for all channels at the same time
  ‘Capture Volume’ | Manipulate sound capture volume
  ‘Joined Capture Volume’ | Manipulate sound capture volume for all channels at a time

Mixer.getenum()

  For enumerated controls, return the currently selected item and the list of items available.

  Returns a tuple (string, list of strings).

  For example, my soundcard has a Mixer called Mono Output Select. Using amixer, I get:

      $ amixer get "Mono Output Select"
      Simple mixer control 'Mono Output Select',0
        Capabilities: enum
        Items: 'Mix' 'Mic'
        Item0: 'Mix'

  Using `alsaaudio`, one could do:

      >>> import alsaaudio
      >>> m = alsaaudio.Mixer('Mono Output Select')
      >>> m.getenum()
      ('Mix', ['Mix', 'Mic'])

  This method will return an empty tuple if the mixer is not an enumerated control.

Mixer.setenum(index)

  For enumerated controls, sets the currently selected item. index is an index into the list of available enumerated items returned by `getenum()`.

Mixer.getrange(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_RAW)

  Return the volume range of the ALSA mixer controlled by this object. The value is a tuple of integers whose meaning is determined by the units argument.

  The optional pcmtype argument can be either `PCM_PLAYBACK` or `PCM_CAPTURE`, which is relevant if the mixer can control both playback and capture volume. The default value is `PCM_PLAYBACK` if the mixer has playback channels, otherwise it is `PCM_CAPTURE`.

  The optional units argument can be one of `VOLUME_UNITS_PERCENTAGE`, `VOLUME_UNITS_RAW`, or `VOLUME_UNITS_DB`.

Mixer.getvolume(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)

  Returns a list with the current volume settings for each channel. The list elements are integers whose meaning is determined by the units argument.

  The optional pcmtype argument can be either `PCM_PLAYBACK` or `PCM_CAPTURE`, which is relevant if the mixer can control both playback and capture volume. The default value is `PCM_PLAYBACK` if the mixer has playback channels, otherwise it is `PCM_CAPTURE`.

  The optional units argument can be one of `VOLUME_UNITS_PERCENTAGE`, `VOLUME_UNITS_RAW`, or `VOLUME_UNITS_DB`.

Mixer.setvolume(volume, channel=None, pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)

  Change the current volume settings for this mixer. The volume argument is an integer whose meaning is determined by the units argument.

  If the optional argument channel is present, the volume is set only for this channel. This assumes that the mixer can control the volume for the channels independently.

  The optional pcmtype argument can be either `PCM_PLAYBACK` or `PCM_CAPTURE`, which is relevant if the mixer can control both playback and capture volume. The default value is `PCM_PLAYBACK` if the mixer has playback channels, otherwise it is `PCM_CAPTURE`.

  The optional units argument can be one of `VOLUME_UNITS_PERCENTAGE`, `VOLUME_UNITS_RAW`, or `VOLUME_UNITS_DB`.

Mixer.getmute()

  Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted.

  This method will fail if the mixer has no playback switch capabilities.

Mixer.setmute(mute[, channel])

  Sets the mute flag to a new value. The mute argument is either 0 for not muted, or 1 for muted.

  The optional channel argument controls which channel is muted. The default is to set the mute flag for all channels.

  This method will fail if the mixer has no playback mute capabilities

Mixer.getrec()

  Return a list indicating the current record mute setting for each channel. 0 means not recording, 1 means recording.

  This method will fail if the mixer has no capture switch capabilities.

Mixer.setrec(capture[, channel])

  Sets the capture mute flag to a new value. The capture argument is either 0 for no capture, or 1 for capture.

  The optional channel argument controls which channel is changed. The default is to set the capture flag for all channels.

  This method will fail if the mixer has no capture switch capabilities.

Mixer.polldescriptors()

  Returns a list of tuples of (file descriptor, eventmask) that can be used to wait for changes on the mixer with select.poll.

  The eventmask value is compatible with `poll.register`__ in the Python `select` module.

Mixer.handleevents()

  Acknowledge events on the `polldescriptors()` file descriptors to prevent subsequent polls from returning the same events again. Returns the number of events that were acknowledged.

Mixer.close()

  Closes the Mixer device.

A rant on the ALSA Mixer API

The ALSA mixer API is extremely complicated - and hardly documented at all. `alsaaudio` implements a much simplified way to access this API. In designing the API I’ve had to make some choices which may limit what can and cannot be controlled through the API. However, if I had chosen to implement the full API, I would have reexposed the horrible complexity/documentation ratio of the underlying API. At least the `alsaaudio` API is easy to understand and use.

If my design choises prevents you from doing something that the underlying API would have allowed, please let me know, so I can incorporate these needs into future versions.

If the current state of affairs annoys you, the best you can do is to write a HOWTO on the API and make this available on the net. Until somebody does this, the availability of ALSA mixer capable devices will stay quite limited.

Unfortunately, I’m not able to create such a HOWTO myself, since I only understand half of the API, and that which I do understand has come from a painful trial and error process.

## Examples

The following example are provided:

- playwav.py

- recordtest.py

- playbacktest.py

- mixertest.py

All examples (except mixertest.py) accept the commandline option -c <cardname>.

To determine a valid card name, use the commandline ALSA player:

    $ aplay -L

or:

    $ python

    >>> import alsaaudio
    >>> alsaaudio.pcms()

mixertest.py accepts the commandline options -d <device> and -c <cardindex>.

### playwav.py

playwav.py plays a wav file.

To test PCM playback (on your default soundcard), run:

    $ python playwav.py <wav file>

### recordtest.py and playbacktest.py

recordtest.py and playbacktest.py will record and play a raw sound file in CD quality.

To test PCM recordings (on your default soundcard), run:

    $ python recordtest.py <filename>

Speak into the microphone, and interrupt the recording at any time with `Ctl-C`.

Play back the recording with:

    $ python playbacktest.py <filename>

### mixertest.py

Without arguments, mixertest.py will list all available controls on the default soundcard.

The output might look like this:

    $ ./mixertest.py
    Available mixer controls:
       'Master'
       'Master Mono'
       'Headphone'
       'PCM'
       'Line'
       'Line In->Rear Out'
       'CD'
       'Mic'
       'PC Speaker'
       'Aux'
       'Mono Output Select'
       'Capture'
       'Mix'
       'Mix Mono'

With a single argument - the control, it will display the settings of that control; for example:

    $ ./mixertest.py Master
    Mixer name: 'Master'
    Capabilities: Playback Volume Playback Mute
    Channel 0 volume: 61%
    Channel 1 volume: 61%

With two arguments, the control and a parameter, it will set the parameter on the mixer:

    $ ./mixertest.py Master mute

This will mute the Master mixer.

Or:

    $ ./mixertest.py Master 40

This sets the volume to 40% on all channels.

To select a different soundcard, use either the device or cardindex argument:

    $ ./mixertest.py -c 0 Master
    Mixer name: 'Master'
    Capabilities: Playback Volume Playback Mute
    Channel 0 volume: 61%
    Channel 1 volume: 61%