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Decided that getenum should return the selected item and the available

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items.

Argument parsing errors are reported with the methodname (minor 
improvement).

Smallish documentation improvements.


git-svn-id: svn://svn.code.sf.net/p/pyalsaaudio/code/trunk@23 ec2f30ec-7544-0410-870e-f70ca00c83f0
This commit is contained in:
larsimmisch
2008-05-21 14:06:26 +00:00
parent 40c4386803
commit df89c12581
19 changed files with 585 additions and 354 deletions
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201
doc/pcm-objects.html
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@@ -56,13 +56,15 @@
</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.
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:
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
(probably a microphone). The PCM object constructor takes the following
arguments:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
@@ -74,18 +76,21 @@ constructor takes the following arguments:
<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.
<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.
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 through
this API yet.
<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
<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:
@@ -108,7 +113,7 @@ PCM objects have the following methods:
<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.
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
</dl>
<p>
@@ -116,7 +121,8 @@ Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
<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
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC,
or PCM_NORMAL
</dl>
<p>
@@ -124,7 +130,7 @@ Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL
<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.
Return the name of the sound card used by this PCM object.
</dl>
<p>
@@ -132,8 +138,9 @@ Return the name of the sound card used by this PCM object.
<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
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>
@@ -141,17 +148,18 @@ and 6 = full 6 channel audio. Few sound cards support more than 2 channels
<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
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>
<td><var>format</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.
The sound <var>format</var> 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:
@@ -168,47 +176,66 @@ The following formats are provided by ALSA:
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<td class="left" >9600 bits/s constant rate encoding for speech</td></tr></tbody>
</table></div>
<p>
@@ -219,9 +246,10 @@ The following formats are provided by ALSA:
<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)
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>
@@ -229,14 +257,16 @@ it may return nothing at all)
<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.
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 number of frames of 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.
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>
@@ -244,50 +274,67 @@ has become available since the last call to read.
<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.
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
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.
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.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-16' xml:id='l2h-16' class="method">pause</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>enable=1</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
If <var>enable</var> is 1, playback or capture is paused. If <var>enable</var> is 0,
playback/capture is resumed.
</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.
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).
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.
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.
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 APIs 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.
Also note, that most timer APIs 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>
@@ -324,7 +371,7 @@ extra writes as nessecary.
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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