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r1274 | casper | 2005-03-26 00:37:10 +0100 (Sat, 26 Mar 2005) | 2 lines Module documentation git-svn-id: svn://svn.code.sf.net/p/pyalsaaudio/code/trunk@9 ec2f30ec-7544-0410-870e-f70ca00c83f0
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328 lines
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<title>4.2 PCM Objects</title>
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<td><a rel="prev" title="4.1 PCM Terminology and"
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<H2><A NAME="SECTION002420000000000000000"> </A>
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<BR>
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4.2 PCM Objects
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</H2>
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<P>
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The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA
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and many other places to handle playback and capture of sampled sound data.
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<P>
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PCM objects in <tt class="module">alsaaudio</tt> are used to do exactly that, either play sample based
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sound or capture sound from some input source (perhaps a microphone). The PCM object
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constructor takes the following arguments:
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><span class="typelabel">class</span> <a name="l2h-6"><tt class="class">PCM</tt></a></b>(</nobr></td>
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<td><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>)</td></tr></table>
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<dd>
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<P>
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<var>type</var> - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
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<P>
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<var>mode</var> - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).
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In PCM_NONBLOCK mode, calls to read will return immediately independent of wether
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there is any actual data to read. Similarly, write calls will return immediately
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without actually writing anything to the playout buffer if the buffer is full.
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<P>
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In the current version of <tt class="module">alsaaudio</tt> PCM_ASYNC is useless, since it relies
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on a callback procedure, which can't be specified from Python.
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<P>
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<var>cardname</var> - specifies which card should be used (this is only relevant
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if you have more than one sound card). Omit to use the default sound card
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<P>
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This will construct a PCM object with default settings:
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<P>
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Sample format: PCM_FORMAT_S16_LE
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<BR>
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Rate: 8000 Hz
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<BR>
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Channels: 2
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<BR>
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Period size: 32 frames
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<BR></dl>
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<P>
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PCM objects have the following methods:
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-7"><tt class="method">pcmtype</tt></a></b>(</nobr></td>
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<td>)</td></tr></table>
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<dd>
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Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-8"><tt class="method">pcmmode</tt></a></b>(</nobr></td>
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<td>)</td></tr></table>
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<dd>
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Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-9"><tt class="method">cardname</tt></a></b>(</nobr></td>
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<td>)</td></tr></table>
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<dd>
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Return the name of the sound card used by this PCM object.
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-10"><tt class="method">setchannels</tt></a></b>(</nobr></td>
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<td><var>nchannels</var>)</td></tr></table>
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<dd>
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Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo,
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and 6 = full 6 channel audio. Few sound cards support more than 2 channels
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-11"><tt class="method">setrate</tt></a></b>(</nobr></td>
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<td><var>rate</var>)</td></tr></table>
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<dd>
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Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality),
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and 96000
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-12"><tt class="method">setformat</tt></a></b>(</nobr></td>
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<td>)</td></tr></table>
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<dd>
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The sound format of the device. Sound format controls how the PCM device interpret data for playback,
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and how data is encoded in captures.
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<P>
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The following formats are provided by ALSA:
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<table border align="center" style="border-collapse: collapse">
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<thead>
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<tr class="tableheader">
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<th align="left"><b>Format</b> </th>
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<th align="left"><b>Description</b> </th>
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</tr>
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</thead>
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<tbody valign="baseline">
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S8</Formats></td>
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<td align="left">Signed 8 bit samples for each channel</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U8</Formats></td>
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<td align="left">Signed 8 bit samples for each channel</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S16_LE</Formats></td>
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<td align="left">Signed 16 bit samples for each channel (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S16_BE</Formats></td>
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<td align="left">Signed 16 bit samples for each channel (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U16_LE</Formats></td>
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<td align="left">Unsigned 16 bit samples for each channel (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U16_BE</Formats></td>
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<td align="left">Unsigned 16 bit samples for each channel (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S24_LE</Formats></td>
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<td align="left">Signed 24 bit samples for each channel (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S24_BE</Formats></td>
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<td align="left">Signed 24 bit samples for each channel (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U24_LE</Formats></td>
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<td align="left">Unsigned 24 bit samples for each channel (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U24_BE</Formats></td>
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<td align="left">Unsigned 24 bit samples for each channel (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S32_LE</Formats></td>
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<td align="left">Signed 32 bit samples for each channel (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_S32_BE</Formats></td>
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<td align="left">Signed 32 bit samples for each channel (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U32_LE</Formats></td>
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<td align="left">Unsigned 32 bit samples for each channel (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_U32_BE</Formats></td>
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<td align="left">Unsigned 32 bit samples for each channel (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT_LE</Formats></td>
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<td align="left">32 bit samples encoded as float. (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT_BE</Formats></td>
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<td align="left">32 bit samples encoded as float (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT64_LE</Formats></td>
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<td align="left">64 bit samples encoded as float. (Little Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_FLOAT64_BE</Formats></td>
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<td align="left">64 bit samples encoded as float. (Big Endian byte order)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_MU_LAW</Formats></td>
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<td align="left">A logarithmic encoding (used by Sun .au files)</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_A_LAW</Formats></td>
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<td align="left">Another logarithmic encoding</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_IMA_ADPCM</Formats></td>
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<td align="left">a 4:1 compressed format defined by the Interactive Multimedia Association</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_MPEG</Formats></td>
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<td align="left">MPEG encoded audio?</td>
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<tr><td align="left" valign="baseline"><Formats>PCM_FORMAT_GSM</Formats></td>
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<td align="left">9600 constant rate encoding well suitet for speech</td></tbody>
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</table>
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<P>
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-13"><tt class="method">setperiodsize</tt></a></b>(</nobr></td>
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<td><var>period</var>)</td></tr></table>
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<dd>
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Sets the actual period size in frames. Each write should consist of exactly this number of frames, and
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each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case
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it may return nothing at all)
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-14"><tt class="method">read</tt></a></b>(</nobr></td>
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<td>)</td></tr></table>
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<dd>
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In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a
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tuple (length,data) where <i>length</i> is the size in bytes of the captured data, and <i>data</i>
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is the captured sound frames as a string. The length of the returned data will be periodsize*framesize
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bytes.
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<P>
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In PCM_NONBLOCK mode, the call will not block, but will return <code>(0,'')</code> if no new period
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has become available since the last call to read.
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</dl>
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<P>
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<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
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<td><nobr><b><a name="l2h-15"><tt class="method">write</tt></a></b>(</nobr></td>
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<td><var>data</var>)</td></tr></table>
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<dd>
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Writes (plays) the sound in data. The length of data <i>must</i> be a multiple of the frame size, and
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<i>should</i> be exactly the size of a period. If less than 'period size' frames are provided, the actual
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playout will not happen until more data is written.
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<P>
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If the device is not in PCM_NONBLOCK mode, this call will block if the kernel buffer is full, and
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until enough sound has been played to allow the sound data to be buffered. The call always returns
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the size of the data provided
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<P>
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In PCM_NONBLOCK mode, the call will return immediately, with a return value of zero, if the buffer is
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full. In this case, the data should be written at a later time.
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<P>
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</dl>
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<P>
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<b>A few hints on using PCM devices for playback</b>
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<P>
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The most common reason for problems with playback of PCM audio, is that the people don't properly understand
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that writes to PCM devices must match <i>exactly</i> the data rate of the device.
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<P>
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If too little data is written to the device will an underrun, and ugly clicking sounds will occur. Conversely,
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of too much data is written to the device, the write function will either block (PCM_NORMAL mode) or return zero
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(PCM_NONBLOCK mode).
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<P>
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If your program does nothing, but play sound, the easiest way is to put the device in PCM_NORMAL mode, and just
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write as much data to the device as possible. This strategy can also be achieved by using a separate thread
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with the sole task of playing out sound.
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<P>
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In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few
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periods by calling write a couple of times, and then use some timer method to write one period size of data to
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the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer
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doesn't expire exactly on time.
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<P>
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Also note, that most timer API's that you can find for Python will cummulate time delays: If you set the timer
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to expire after 1/10'th of a second, the actual timeout will happen slightly later, which will accumulate to
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quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add
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extra writes as nessecary.
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<P>
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<DIV CLASS="navigation">
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<p><hr>
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border='0' height='32' alt='Previous Page' width='32'></A></td>
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href="module-alsaaudio.html"><img src='up.gif'
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border='0' height='32' alt='Up One Level' width='32'></A></td>
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border='0' height='32' alt='Next Page' width='32'></A></td>
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<td align="center" width="100%">PyAlsaAudio</td>
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<td><a rel="contents" title="Table of Contents"
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rel="contents" title="Table of Contents"
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href="contents.html"><img src='contents.gif'
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border='0' height='32' alt='Contents' width='32'></A></td>
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border='0' height='32' alt='' width='32'></td>
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<b class="navlabel">Previous:</b>
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<a class="sectref" rel="prev" HREF="node7.html">4.1 PCM Terminology and</A>
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<b class="navlabel">Up:</b>
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<a class="sectref" rel="parent" href="module-alsaaudio.html">4 alsaaudio</A>
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<b class="navlabel">Next:</b>
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<a class="sectref" rel="next" href="mixer-objects.html">4.3 Mixer Objects</A>
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<span class="release-info">Release 0.1.</span>
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