Platforms: Linux
The alsaaudio module defines functions and classes for using ALSA.
List the available cards by name (suitable for PCM objects).
List the available mixers. The optional cardindex specifies which card should be queried. The default is 0.
This class is used to represent a PCM device (both for playback and recording - capture). The arguments are:
This class is used to access a specific ALSA mixer. The arguments are:
Exception raised when an operation fails for a ALSA specific reason. The exception argument is a string describing the reason of the failure.
PCM objects in alsaaudio can play or capture (record) PCM sound through speakers or a microphone. The PCM constructor takes the following arguments:
type - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
mode - can be either PCM_NONBLOCK, or PCM_NORMAL (the default). In PCM_NONBLOCK mode, calls to read() will return immediately independent of whether there is any actual data to read. Similarly, calls to write() will return immediately without actually writing anything to the playout buffer if the buffer is full [1].
card - specifies which card should be used. This can be a string like ‘default’ or a name that was returned from the cards() function.
This will construct a PCM object with these default settings:
PCM objects have the following methods:
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL
Return the name of the sound card used by this PCM object.
Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo, and 6 = full 6 channel audio. Few sound cards support more than 2 channels
Set the sample rate in Hz for the device. Typical values are 8000 (mainly used for telephony), 16000, 44100 (CD quality), and 96000.
The sound format of the device. Sound format controls how the PCM device interpret data for playback, and how data is encoded in captures.
The following formats are provided by ALSA:
| Format | Description |
|---|---|
| PCM_FORMAT_S8 | Signed 8 bit samples for each channel |
| PCM_FORMAT_U8 | Signed 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) |
| PCM_FORMAT_S24_BE | Signed 24 bit samples for each channel (Big Endian byte order)} |
| PCM_FORMAT_U24_LE | Unsigned 24 bit samples for each channel (Little Endian byte order) |
| PCM_FORMAT_U24_BE | Unsigned 24 bit samples for each channel (Big Endian byte order) |
| 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 |
Sets the actual period size in frames. Each write should consist of exactly this number of frames, and each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case it may return nothing at all)
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.
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.
If enable is 1, playback or capture is paused. If enable is 0, playback/capture is resumed.
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 provides access to the ALSA mixer API.
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 include ‘Master Mono’, ‘PCM’, ‘Line’, etc.
id - the id of the mixer control. Default is 0
cardindex - specifies which card should be used [3]. 0 is the first sound card.
Note: For a list of available controls, you can also use amixer:
amixer
Mixer objects have the following methods:
Return the name of the sound card used by this Mixer object
Return the name of the specific mixer controlled by this object, For example ‘Master’ or ‘PCM’
Return the ID of the ALSA mixer controlled by this object.
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
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 |
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.
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.
Return the volume range of the ALSA mixer controlled by this object.
The optional direction argument can be either ‘playback’ or ‘capture’, which is relevant if the mixer can control both playback and capture volume. The default value is ‘playback’ if the mixer has this capability, otherwise ‘capture’
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.
Returns a list with the current volume settings for each channel. The list elements are integer percentages.
The optional direction argument can be either ‘playback’ or ‘capture’, which is relevant if the mixer can control both playback and capture volume. The default value is ‘playback’ if the mixer has this capability, otherwise ‘capture’
Change the current volume settings for this mixer. The volume argument controls the new volume setting as an integer percentage.
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 direction argument can be either ‘playback’ or ‘capture’ is relevant if the mixer has independent playback and capture volume capabilities, and controls which of the volumes if changed. The default is ‘playback’ if the mixer has this capability, otherwise ‘capture’.
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
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.
Returns a tuple of (file descriptor, eventmask) that can be used to wait for changes on the mixer with select.poll.
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.
The following example are provided:
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.cards()
mixertest.py accepts the commandline option -c <cardindex>. Card indices start at 0.
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 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>
Without arguments, mixertest.py will list all available controls. 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.
Footnotes
| [1] | ALSA also allows PCM_ASYNC, but this is not supported yet. |
| [2] | alsaaudio will leave any name alone that has a ‘:’ (colon) in it. |
| [3] | This is inconsistent with the PCM objects, which use names, but it is consistent with aplay and amixer. |