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<H2><A NAME="SECTION002420000000000000000"></A>
<A NAME="pcm-objects"></A>
<BR>
4.2 PCM Objects
</H2>
<P>
The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA
and many other places to handle playback and capture of sampled sound data.
<P>
PCM objects in <tt class="module">alsaaudio</tt> are used to do exactly that, either play sample based
sound or capture sound from some input source (perhaps a microphone). The PCM object
constructor takes the following arguments:
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><span class="typelabel">class</span>&nbsp;<tt id='l2h-6' xml:id='l2h-6' class="class">PCM</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>type</var><big>]</big><var>, </var><big>[</big><var>mode</var><big>]</big><var>, </var><big>[</big><var>cardname</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
<P>
<var>type</var> - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
<P>
<var>mode</var> - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).
In PCM_NONBLOCK mode, calls to read will return immediately independent of wether
there is any actual data to read. Similarly, write calls will return immediately
without actually writing anything to the playout buffer if the buffer is full.
<P>
In the current version of <tt class="module">alsaaudio</tt> PCM_ASYNC is useless, since it relies
on a callback procedure, which can't be specified from Python.
<P>
<var>cardname</var> - specifies which card should be used (this is only relevant
if you have more than one sound card). Omit to use the default sound card
<P>
This will construct a PCM object with default settings:
<P>
Sample format: PCM_FORMAT_S16_LE
<BR>
Rate: 8000 Hz
<BR>
Channels: 2
<BR>
Period size: 32 frames
<BR></dl>
<P>
PCM objects have the following methods:
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-7' xml:id='l2h-7' class="method">pcmtype</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-8' xml:id='l2h-8' class="method">pcmmode</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-9' xml:id='l2h-9' class="method">cardname</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the name of the sound card used by this PCM object.
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-10' xml:id='l2h-10' class="method">setchannels</tt></b>(</nobr></td>
<td><var>nchannels</var>)</td></tr></table></dt>
<dd>
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
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-11' xml:id='l2h-11' class="method">setrate</tt></b>(</nobr></td>
<td><var>rate</var>)</td></tr></table></dt>
<dd>
Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality),
and 96000
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-12' xml:id='l2h-12' class="method">setformat</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
The sound format of the device. Sound format controls how the PCM device interpret data for playback,
and how data is encoded in captures.
<P>
The following formats are provided by ALSA:
<div class="center"><table class="realtable">
<thead>
<tr>
<th class="left" >Format</th>
<th class="left" >Description</th>
</tr>
</thead>
<tbody>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S8</Formats></td>
<td class="left" >Signed 8 bit samples for each channel</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U8</Formats></td>
<td class="left" >Signed 8 bit samples for each channel</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S16_LE</Formats></td>
<td class="left" >Signed 16 bit samples for each channel (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S16_BE</Formats></td>
<td class="left" >Signed 16 bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U16_LE</Formats></td>
<td class="left" >Unsigned 16 bit samples for each channel (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U16_BE</Formats></td>
<td class="left" >Unsigned 16 bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S24_LE</Formats></td>
<td class="left" >Signed 24 bit samples for each channel (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S24_BE</Formats></td>
<td class="left" >Signed 24 bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U24_LE</Formats></td>
<td class="left" >Unsigned 24 bit samples for each channel (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U24_BE</Formats></td>
<td class="left" >Unsigned 24 bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S32_LE</Formats></td>
<td class="left" >Signed 32 bit samples for each channel (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_S32_BE</Formats></td>
<td class="left" >Signed 32 bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U32_LE</Formats></td>
<td class="left" >Unsigned 32 bit samples for each channel (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_U32_BE</Formats></td>
<td class="left" >Unsigned 32 bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT_LE</Formats></td>
<td class="left" >32 bit samples encoded as float. (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT_BE</Formats></td>
<td class="left" >32 bit samples encoded as float (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT64_LE</Formats></td>
<td class="left" >64 bit samples encoded as float. (Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT64_BE</Formats></td>
<td class="left" >64 bit samples encoded as float. (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_MU_LAW</Formats></td>
<td class="left" >A logarithmic encoding (used by Sun .au files)</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_A_LAW</Formats></td>
<td class="left" >Another logarithmic encoding</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_IMA_ADPCM</Formats></td>
<td class="left" >a 4:1 compressed format defined by the Interactive Multimedia Association</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_MPEG</Formats></td>
<td class="left" >MPEG encoded audio?</td></tr>
<tr><td class="left" valign="baseline"><Formats>PCM_FORMAT_GSM</Formats></td>
<td class="left" >9600 constant rate encoding well suitet for speech</td></tr></tbody>
</table></div>
<P>
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-13' xml:id='l2h-13' class="method">setperiodsize</tt></b>(</nobr></td>
<td><var>period</var>)</td></tr></table></dt>
<dd>
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)
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-14' xml:id='l2h-14' class="method">read</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a
tuple (length,data) where <em>length</em> is the size in bytes of the captured data, and <em>data</em>
is the captured sound frames as a string. The length of the returned data will be periodsize*framesize
bytes.
<P>
In PCM_NONBLOCK mode, the call will not block, but will return <code>(0,'')</code> if no new period
has become available since the last call to read.
</dl>
<P>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-15' xml:id='l2h-15' class="method">write</tt></b>(</nobr></td>
<td><var>data</var>)</td></tr></table></dt>
<dd>
Writes (plays) the sound in data. The length of data <em>must</em> be a multiple of the frame size, and
<em>should</em> 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.
<P>
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
<P>
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.
<P>
</dl>
<P>
<strong>A few hints on using PCM devices for playback</strong>
<P>
The most common reason for problems with playback of PCM audio, is that the people don't properly understand
that writes to PCM devices must match <em>exactly</em> the data rate of the device.
<P>
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).
<P>
If your program does nothing, but play sound, the easiest way 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.
<P>
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.
<P>
Also note, that most timer API's that you can find for Python will cummulate 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.
<P>
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