auracaster/sw_pyalsaaudio
3
0
Fork 0
forked from auracaster/pyalsaaudio
Code Pull Requests Activity

Ported to Python3.0

Browse Source 
Converted documentation to Sphinx.

Added playwav.py



git-svn-id: svn://svn.code.sf.net/p/pyalsaaudio/code/trunk@25 ec2f30ec-7544-0410-870e-f70ca00c83f0
This commit is contained in:
larsimmisch
2008-11-25 20:12:56 +00:00
parent 7fa6f1c85c
commit 5c2a006553
34 changed files with 952 additions and 1439 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
CHANGES
View File

@@ -1,6 +1,8 @@
Version 0.4:
- Support for Python 3.0
- Documentation in reStructuredText; use Sphinx instead of LaTeX.
- added mixer.getenum()
- small documentation improvements
Version 0.3:
- wrapped blocking calls with Py_BEGIN_ALLOW_THREADS/Py_END_ALLOW_THREADS

2
README
View File

@@ -41,7 +41,7 @@ And then as root:
Using the API
=============
There is a reasonably usefull API documentation included in the module
There is a reasonably useful API documentation included in the module
documentation, which can be found in the doc subdirectory of the source
distribution.

158
alsaaudio.c
View File

@@ -5,7 +5,7 @@
* The standard audio API for Linux since kernel 2.6
*
* Contributed by Unispeed A/S (http://www.unispeed.com)
* Author: Casper Wilstup (cwi@unispeed.dk)
* Author: Casper Wilstup (cwi@aves.dk)
*
* Bug fixes and maintenance by Lars Immisch <lars@ibp.de>
*
@@ -14,7 +14,10 @@
*/
#include "Python.h"
#if PY_MAJOR_VERSION < 3
#include "stringobject.h"
#define PyUnicode_FromString PyString_FromString
#endif
#include <alsa/asoundlib.h>
#include <stdio.h>
@@ -237,8 +240,8 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
val = snd_pcm_hw_params_get_sbits(hwparams);
printf("significant bits = %d\n", val);
snd_pcm_hw_params_get_tick_time(hwparams, &val, &dir);
printf("tick time = %d us\n", val);
snd_pcm_hw_params_get_period_time(hwparams, &val, &dir);
printf("period time = %d us\n", val);
val = snd_pcm_hw_params_is_batch(hwparams);
printf("is batch = %d\n", val);
@@ -305,7 +308,7 @@ Returns the mode of the PCM object. One of:\n\
static PyObject *
alsapcm_cardname(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":cardname")) return NULL;
return PyString_FromString(self->cardname);
return PyUnicode_FromString(self->cardname);
}
PyDoc_STRVAR(cardname_doc,
@@ -429,12 +432,16 @@ alsapcm_read(alsapcm_t *self, PyObject *args) {
res = 0;
}
else if (res < 0) {
PyErr_SetString(ALSAAudioError,snd_strerror(res));
PyErr_SetString(ALSAAudioError, snd_strerror(res));
return NULL;
}
}
return Py_BuildValue("is#",res,buffer,res*self->framesize);
#if PY_MAJOR_VERSION < 3
return Py_BuildValue("is#", res, buffer, res*self->framesize);
#else
return Py_BuildValue("iy#", res, buffer, res*self->framesize);
#endif
}
PyDoc_STRVAR(read_doc,
@@ -443,19 +450,33 @@ PyDoc_STRVAR(read_doc,
In PCM_NORMAL mode, this function blocks until a full period is\n\
available, and then returns a tuple (length,data) where length is\n\
the number of frames of the captured data, and data is the captured sound\n\
frames as a string. The length of the returned data will be\n\
periodsize*framesize bytes.\n\
frames as bytes (or a string in Python 2.x). The length of the returned data\n\
will be periodsize*framesize bytes.\n\
\n\
In PCM_NONBLOCK mode, the call will not block, but will return (0,'')\n\
if no new period has become available since the last call to read.");
static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) {
char *data;
int datalen;
int res;
if (!PyArg_ParseTuple(args,"s#:write",&data,&datalen)) return NULL;
if (datalen%self->framesize) {
int datalen;
char *data;
#if PY_MAJOR_VERSION < 3
if (!PyArg_ParseTuple(args,"s#:write",&data,&datalen))
return NULL;
#else
Py_buffer buf;
if (!PyArg_ParseTuple(args,"y*:write",&buf))
return NULL;
data = buf.buf;
datalen = buf.len;
#endif
if (datalen % self->framesize) {
PyErr_SetString(ALSAAudioError,
"Data size must be a multiple of framesize");
return NULL;
@@ -548,8 +569,12 @@ alsapcm_getattr(alsapcm_t *self, char *name) {
#endif
static PyTypeObject ALSAPCMType = {
#if PY_MAJOR_VERSION < 3
PyObject_HEAD_INIT(&PyType_Type)
0, /* ob_size */
#else
PyVarObject_HEAD_INIT(&PyType_Type, 0)
#endif
"alsaaudio.PCM", /* tp_name */
sizeof(alsapcm_t), /* tp_basicsize */
0, /* tp_itemsize */
@@ -647,7 +672,7 @@ alsamixer_list(PyObject *self, PyObject *args) {
{
PyObject *mixer;
snd_mixer_selem_get_id(elem, sid);
mixer = PyString_FromString(snd_mixer_selem_id_get_name(sid));
mixer = PyUnicode_FromString(snd_mixer_selem_id_get_name(sid));
PyList_Append(result,mixer);
Py_DECREF(mixer);
}
@@ -788,7 +813,7 @@ static void alsamixer_dealloc(alsamixer_t *self) {
static PyObject *
alsamixer_cardname(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":cardname")) return NULL;
return PyString_FromString(self->cardname);
return PyUnicode_FromString(self->cardname);
}
PyDoc_STRVAR(mixer_cardname_doc,
@@ -800,7 +825,7 @@ Returns the name of the sound card used by this Mixer object.");
static PyObject *
alsamixer_mixer(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":mixer")) return NULL;
return PyString_FromString(self->controlname);
return PyUnicode_FromString(self->controlname);
}
PyDoc_STRVAR(mixer_doc,
@@ -828,17 +853,17 @@ alsamixer_volumecap(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":volumecap")) return NULL;
result = PyList_New(0);
if (self->volume_cap&MIXER_CAP_VOLUME)
PyList_Append(result,PyString_FromString("Volume"));
PyList_Append(result,PyUnicode_FromString("Volume"));
if (self->volume_cap&MIXER_CAP_VOLUME_JOINED)
PyList_Append(result,PyString_FromString("Joined Volume"));
PyList_Append(result,PyUnicode_FromString("Joined Volume"));
if (self->volume_cap&MIXER_CAP_PVOLUME)
PyList_Append(result,PyString_FromString("Playback Volume"));
PyList_Append(result,PyUnicode_FromString("Playback Volume"));
if (self->volume_cap&MIXER_CAP_PVOLUME_JOINED)
PyList_Append(result,PyString_FromString("Joined Playback Volume"));
PyList_Append(result,PyUnicode_FromString("Joined Playback Volume"));
if (self->volume_cap&MIXER_CAP_CVOLUME)
PyList_Append(result,PyString_FromString("Capture Volume"));
PyList_Append(result,PyUnicode_FromString("Capture Volume"));
if (self->volume_cap&MIXER_CAP_CVOLUME_JOINED)
PyList_Append(result,PyString_FromString("Joined Capture Volume"));
PyList_Append(result,PyUnicode_FromString("Joined Capture Volume"));
return result;
}
@@ -862,19 +887,19 @@ alsamixer_switchcap(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":switchcap")) return NULL;
result = PyList_New(0);
if (self->volume_cap&MIXER_CAP_SWITCH)
PyList_Append(result,PyString_FromString("Mute"));
PyList_Append(result,PyUnicode_FromString("Mute"));
if (self->volume_cap&MIXER_CAP_SWITCH_JOINED)
PyList_Append(result,PyString_FromString("Joined Mute"));
PyList_Append(result,PyUnicode_FromString("Joined Mute"));
if (self->volume_cap&MIXER_CAP_PSWITCH)
PyList_Append(result,PyString_FromString("Playback Mute"));
PyList_Append(result,PyUnicode_FromString("Playback Mute"));
if (self->volume_cap&MIXER_CAP_PSWITCH_JOINED)
PyList_Append(result,PyString_FromString("Joined Playback Mute"));
PyList_Append(result,PyUnicode_FromString("Joined Playback Mute"));
if (self->volume_cap&MIXER_CAP_CSWITCH)
PyList_Append(result,PyString_FromString("Capture Mute"));
PyList_Append(result,PyUnicode_FromString("Capture Mute"));
if (self->volume_cap&MIXER_CAP_CSWITCH_JOINED)
PyList_Append(result,PyString_FromString("Joined Capture Mute"));
PyList_Append(result,PyUnicode_FromString("Joined Capture Mute"));
if (self->volume_cap&MIXER_CAP_CSWITCH_EXCLUSIVE)
PyList_Append(result,PyString_FromString("Capture Exclusive"));
PyList_Append(result,PyUnicode_FromString("Capture Exclusive"));
return result;
}
@@ -1064,7 +1089,7 @@ alsamixer_getenum(alsamixer_t *self, PyObject *args) {
return NULL;
}
PyTuple_SetItem(result, 0, PyString_FromString(name));
PyTuple_SetItem(result, 0, PyUnicode_FromString(name));
elems = PyList_New(count);
if (!elems)
@@ -1082,7 +1107,7 @@ alsamixer_getenum(alsamixer_t *self, PyObject *args) {
return NULL;
}
PyList_SetItem(elems, i, PyString_FromString(name));
PyList_SetItem(elems, i, PyUnicode_FromString(name));
}
PyTuple_SetItem(result, 1, elems);
@@ -1339,8 +1364,12 @@ alsamixer_getattr(alsapcm_t *self, char *name) {
#endif
static PyTypeObject ALSAMixerType = {
#if PY_MAJOR_VERSION < 3
PyObject_HEAD_INIT(&PyType_Type)
0, /* ob_size */
#else
PyVarObject_HEAD_INIT(&PyType_Type, 0)
#endif
"alsaaudio.Mixer", /* tp_name */
sizeof(alsamixer_t), /* tp_basicsize */
0, /* tp_itemsize */
@@ -1390,32 +1419,73 @@ static PyMethodDef alsaaudio_methods[] = {
{ 0, 0 },
};
#if PY_MAJOR_VERSION >= 3
#define _EXPORT_INT(mod, name, value) \
if (PyModule_AddIntConstant(mod, name, (long) value) == -1) return NULL;
static struct PyModuleDef alsaaudio_module = {
PyModuleDef_HEAD_INIT,
"alsaaudio",
alsaaudio_module_doc,
-1,
alsaaudio_methods,
0, /* m_reload */
0, /* m_traverse */
0, /* m_clear */
0, /* m_free */
};
#else
#define _EXPORT_INT(mod, name, value) \
if (PyModule_AddIntConstant(mod, name, (long) value) == -1) return;
void initalsaaudio(void) {
#endif // 3.0
#if PY_MAJOR_VERSION < 3
void initalsaaudio(void)
#else
PyObject *PyInit_alsaaudio(void)
#endif
{
PyObject *m;
ALSAPCMType.tp_new = alsapcm_new;
ALSAMixerType.tp_new = alsamixer_new;
PyEval_InitThreads();
m = Py_InitModule3("alsaaudio",alsaaudio_methods,alsaaudio_module_doc);
#if PY_MAJOR_VERSION < 3
m = Py_InitModule3("alsaaudio", alsaaudio_methods, alsaaudio_module_doc);
if (!m)
return;
#else
m = PyModule_Create(&alsaaudio_module);
if (!m)
return NULL;
#endif
ALSAAudioError = PyErr_NewException("alsaaudio.ALSAAudioError", NULL, NULL);
if (ALSAAudioError) {
/* Each call to PyModule_AddObject decrefs it; compensate: */
if (!ALSAAudioError)
#if PY_MAJOR_VERSION < 3
return;
#else
return NULL;
#endif
Py_INCREF(&ALSAPCMType);
PyModule_AddObject(m,"PCM",(PyObject *)&ALSAPCMType);
/* Each call to PyModule_AddObject decrefs it; compensate: */
Py_INCREF(&ALSAMixerType);
PyModule_AddObject(m,"Mixer",(PyObject *)&ALSAMixerType);
Py_INCREF(ALSAAudioError);
PyModule_AddObject(m, "ALSAAudioError", ALSAAudioError);
}
Py_INCREF(&ALSAPCMType);
PyModule_AddObject(m, "PCM", (PyObject *)&ALSAPCMType);
Py_INCREF(&ALSAMixerType);
PyModule_AddObject(m, "Mixer", (PyObject *)&ALSAMixerType);
Py_INCREF(ALSAAudioError);
PyModule_AddObject(m, "ALSAAudioError", ALSAAudioError);
_EXPORT_INT(m, "PCM_PLAYBACK",SND_PCM_STREAM_PLAYBACK);
_EXPORT_INT(m, "PCM_CAPTURE",SND_PCM_STREAM_CAPTURE);
@@ -1465,4 +1535,8 @@ void initalsaaudio(void) {
_EXPORT_INT(m, "MIXER_SCHN_REAR_CENTER", SND_MIXER_SCHN_REAR_CENTER);
_EXPORT_INT(m, "MIXER_SCHN_MONO", SND_MIXER_SCHN_MONO);
#endif
#if PY_MAJOR_VERSION >= 3
return m;
#endif
}

77
doc/Makefile Normal file
View File

@@ -0,0 +1,77 @@
# Makefile for Sphinx documentation
#
# You can set these variables from the command line.
SPHINXOPTS =
SPHINXBUILD = sphinx-build
PAPER =
# Internal variables.
PAPEROPT_a4 = -D latex_paper_size=a4
PAPEROPT_letter = -D latex_paper_size=letter
ALLSPHINXOPTS = -d .build/doctrees $(PAPEROPT_$(PAPER)) $(SPHINXOPTS) .
SFUSER = larsimmisch
.PHONY: help clean html web pickle htmlhelp latex changes linkcheck install
all: html
help:
@echo "Please use \`make <target>' where <target> is one of"
@echo " html to make standalone HTML files"
@echo " pickle to make pickle files (usable by e.g. sphinx-web)"
@echo " htmlhelp to make HTML files and a HTML help project"
@echo " latex to make LaTeX files, you can set PAPER=a4 or PAPER=letter"
@echo " changes to make an overview over all changed/added/deprecated items"
@echo " linkcheck to check all external links for integrity"
clean:
-rm -rf .build/*
html:
mkdir -p .build/html .build/doctrees
$(SPHINXBUILD) -b html $(ALLSPHINXOPTS) .build/html
@echo
@echo "Build finished. The HTML pages are in .build/html."
pickle:
mkdir -p .build/pickle .build/doctrees
$(SPHINXBUILD) -b pickle $(ALLSPHINXOPTS) .build/pickle
@echo
@echo "Build finished; now you can process the pickle files or run"
@echo " sphinx-web .build/pickle"
@echo "to start the sphinx-web server."
web: pickle
htmlhelp:
mkdir -p .build/htmlhelp .build/doctrees
$(SPHINXBUILD) -b htmlhelp $(ALLSPHINXOPTS) .build/htmlhelp
@echo
@echo "Build finished; now you can run HTML Help Workshop with the" \
".hhp project file in .build/htmlhelp."
latex:
mkdir -p .build/latex .build/doctrees
$(SPHINXBUILD) -b latex $(ALLSPHINXOPTS) .build/latex
@echo
@echo "Build finished; the LaTeX files are in .build/latex."
@echo "Run \`make all-pdf' or \`make all-ps' in that directory to" \
"run these through (pdf)latex."
changes:
mkdir -p .build/changes .build/doctrees
$(SPHINXBUILD) -b changes $(ALLSPHINXOPTS) .build/changes
@echo
@echo "The overview file is in .build/changes."
linkcheck:
mkdir -p .build/linkcheck .build/doctrees
$(SPHINXBUILD) -b linkcheck $(ALLSPHINXOPTS) .build/linkcheck
@echo
@echo "Link check complete; look for any errors in the above output " \
"or in .build/linkcheck/output.txt."
install:
scp -r .build/html/* $(SFUSER),pyalsaaudio@web.sourceforge.net:htdocs

111
doc/about.html
View File

@@ -1,111 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<link rel="STYLESHEET" href="pyalsaaudio.css" type='text/css' />
<link rel="first" href="pyalsaaudio.html" title='PyAlsaAudio' />
<link rel='contents' href='contents.html' title="Contents" />
<link rel='last' href='about.html' title='About this document...' />
<link rel='help' href='about.html' title='About this document...' />
<link rel="prev" href="contents.html" />
<link rel="parent" href="pyalsaaudio.html" />
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name='aesop' content='information' />
<title>About this document ...</title>
</head>
<body>
<div class="navigation">
<div id='top-navigation-panel' xml:id='top-navigation-panel'>
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="4.4 alsa Examples"
href="pcm-example.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
<td class='online-navigation'><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Previous:</b>
<a class="sectref" rel="prev" href="pcm-example.html">4.4 ALSA Examples</a>
<b class="navlabel">Up:</b>
<a class="sectref" rel="parent" href="pyalsaaudio.html">PyAlsaAudio</a>
</div>
<hr /></div>
</div>
<!--End of Navigation Panel-->
<h1><a name="SECTION003000000000000000000">
About this document ...</a>
</h1>
<strong>PyAlsaAudio</strong>
<p> This document was generated using the <a
href="http://saftsack.fs.uni-bayreuth.de/~latex2ht/">
<strong>LaTeX</strong>2<tt>HTML</tt></a> translator.
</p>
<p> <a
href="http://saftsack.fs.uni-bayreuth.de/~latex2ht/">
<strong>LaTeX</strong>2<tt>HTML</tt></a> is Copyright &copy;
1993, 1994, 1995, 1996, 1997, <a
href="http://cbl.leeds.ac.uk/nikos/personal.html">Nikos
Drakos</a>, Computer Based Learning Unit, University of
Leeds, and Copyright &copy; 1997, 1998, <a
href="http://www.maths.mq.edu.au/~ross/">Ross
Moore</a>, Mathematics Department, Macquarie University,
Sydney.
</p>
<p> The application of <a
href="http://saftsack.fs.uni-bayreuth.de/~latex2ht/">
<strong>LaTeX</strong>2<tt>HTML</tt></a> to the Python
documentation has been heavily tailored by Fred L. Drake,
Jr. Original navigation icons were contributed by Christopher
Petrilli.
</p>
<div class="navigation">
<div class='online-navigation'>
<p></p><hr />
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="4.4 alsa Examples"
href="pcm-example.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
<td class='online-navigation'><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Previous:</b>
<a class="sectref" rel="prev" href="pcm-example.html">4.4 ALSA Examples</a>
<b class="navlabel">Up:</b>
<a class="sectref" rel="parent" href="pyalsaaudio.html">PyAlsaAudio</a>
</div>
</div>
<hr />
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->
</body>
</html>

BIN
doc/blank.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 1.9 KiB

BIN
doc/blank.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 1.0 KiB

BIN
doc/contents.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 438 B

107
doc/contents.html
View File

@@ -1,107 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<link rel="STYLESHEET" href="pyalsaaudio.css" type='text/css' />
<link rel="first" href="pyalsaaudio.html" title='PyAlsaAudio' />
<link rel='contents' href='contents.html' title="Contents" />
<link rel='last' href='about.html' title='About this document...' />
<link rel='help' href='about.html' title='About this document...' />
<link rel="next" href="about.html" />
<link rel="prev" href="front.html" />
<link rel="parent" href="pyalsaaudio.html" />
<link rel="next" href="node3.html" />
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name='aesop' content='information' />
<title>Contents</title>
</head>
<body>
<div class="navigation">
<div id='top-navigation-panel' xml:id='top-navigation-panel'>
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="Front Matter"
href="front.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
<td class='online-navigation'><a rel="next" title="1 what is ALSA"
href="node3.html"><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></a></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Previous:</b>
<a class="sectref" rel="prev" href="front.html">Front Matter</a>
<b class="navlabel">Up:</b>
<a class="sectref" rel="parent" href="pyalsaaudio.html">PyAlsaAudio</a>
<b class="navlabel">Next:</b>
<a class="sectref" rel="next" href="node3.html">1 What is ALSA</a>
</div>
<hr /></div>
</div>
<!--End of Navigation Panel-->
<br><h2><a name="SECTION002000000000000000000">
Contents</a>
</h2>
<!--Table of Contents-->
<ul class="TofC">
<li><ul>
<li><a href="node3.html">1 What is ALSA</a>
<li><a href="node4.html">2 ALSA and Python</a>
<li><a href="node5.html">3 Installation</a>
<li><a href="module-alsaaudio.html">4 alsaaudio</a>
<ul>
<li><a href="node7.html">4.1 PCM Terminology and Concepts</a>
<li><a href="pcm-objects.html">4.2 PCM Objects</a>
<li><a href="mixer-objects.html">4.3 Mixer Objects</a>
<li><a href="pcm-example.html">4.4 ALSA Examples</a>
</ul></ul></ul>
<!--End of Table of Contents-->
<p>
<div class="navigation">
<div class='online-navigation'>
<p></p><hr />
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="Front Matter"
href="front.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
<td class='online-navigation'><a rel="next" title="1 what is ALSA"
href="node3.html"><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></a></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Previous:</b>
<a class="sectref" rel="prev" href="front.html">Front Matter</a>
<b class="navlabel">Up:</b>
<a class="sectref" rel="parent" href="pyalsaaudio.html">PyAlsaAudio</a>
<b class="navlabel">Next:</b>
<a class="sectref" rel="next" href="node3.html">1 What is ALSA</a>
</div>
</div>
<hr />
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->
</body>
</html>

BIN
doc/contents.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 649 B

121
doc/front.html
View File

@@ -1,121 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<link rel="STYLESHEET" href="pyalsaaudio.css" type='text/css' />
<link rel="first" href="pyalsaaudio.html" title='PyAlsaAudio' />
<link rel='contents' href='contents.html' title="Contents" />
<link rel='last' href='about.html' title='About this document...' />
<link rel='help' href='about.html' title='About this document...' />
<link rel="next" href="contents.html" />
<link rel="prev" href="pyalsaaudio.html" />
<link rel="parent" href="pyalsaaudio.html" />
<link rel="next" href="contents.html" />
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name='aesop' content='information' />
<title>Front Matter</title>
</head>
<body>
<div class="navigation">
<div id='top-navigation-panel' xml:id='top-navigation-panel'>
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
<td class='online-navigation'><a rel="next" title="Contents"
href="contents.html"><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></a></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Previous:</b>
<a class="sectref" rel="prev" href="pyalsaaudio.html">PyAlsaAudio</a>
<b class="navlabel">Up:</b>
<a class="sectref" rel="parent" href="pyalsaaudio.html">PyAlsaAudio</a>
<b class="navlabel">Next:</b>
<a class="sectref" rel="next" href="contents.html">Contents</a>
</div>
<hr /></div>
</div>
<!--End of Navigation Panel-->
<h1><a name="SECTION001000000000000000000"></a><a name="front"></a>
<br>
Front Matter
</h1>
<p>
This software is licensed under the PSF license - the same one used
by the majority of the python distribution. Basically you can use it
for anything you wish (even commercial purposes). There is no warranty
whatsoever.
<p>
<h3>Abstract:</h3>
<div class="ABSTRACT">
This package contains wrappers for accessing the ALSA API from Python.
It is currently fairly complete for PCM devices and Mixer access. MIDI
sequencer support is low on my priority list, but volunteers are
welcome.
<p>
If you find bugs in the wrappers please use the SourceForge bug
tracker. Please don't send bug reports regarding ALSA specifically.
There are several bugs in this API, and those should be reported to
the ALSA team - not me.
</div>
<p>
<p>
<div class="navigation">
<div class='online-navigation'>
<p></p><hr />
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="PyAlsaAudio"
href="pyalsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
<td class='online-navigation'><a rel="next" title="Contents"
href="contents.html"><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></a></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Previous:</b>
<a class="sectref" rel="prev" href="pyalsaaudio.html">PyAlsaAudio</a>
<b class="navlabel">Up:</b>
<a class="sectref" rel="parent" href="pyalsaaudio.html">PyAlsaAudio</a>
<b class="navlabel">Next:</b>
<a class="sectref" rel="next" href="contents.html">Contents</a>
</div>
</div>
<hr />
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->
</body>
</html>

BIN
doc/index.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 289 B

118
doc/index.html
View File

@@ -1,118 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<link rel="STYLESHEET" href="pyalsaaudio.css" type='text/css' />
<link rel="first" href="pyalsaaudio.html" title='PyAlsaAudio' />
<link rel='contents' href='contents.html' title="Contents" />
<link rel='last' href='about.html' title='About this document...' />
<link rel='help' href='about.html' title='About this document...' />
<link rel="next" href="front.html" />
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name='aesop' content='information' />
<title>PyAlsaAudio</title>
</head>
<body>
<div class="navigation">
<div id='top-navigation-panel' xml:id='top-navigation-panel'>
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></td>
<td class='online-navigation'><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></td>
<td class='online-navigation'><a rel="next" title="Front Matter"
href="front.html"><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></a></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Next:</b>
<a class="sectref" rel="next" href="front.html">Front Matter</a>
</div>
<hr /></div>
</div>
<!--End of Navigation Panel-->
<div class="titlepage">
<div class='center'>
<h1>PyAlsaAudio</h1>
<p><b><font size="+2">Casper Wilstrup</font></b></p>
<p>cwi@aves.dk</p>
<p></p>
</div>
</div>
<p>
<p><br /></p><hr class='online-navigation' />
<div class='online-navigation'>
<!--Table of Child-Links-->
<ul>
<li><a href="http://sourceforge.net/">SourceForge</li>
<ul>
<li><a href="http://sourceforge.net/projects/pyalsaaudio/">Summary page</a>
<li><a href="http://sourceforge.net/project/showfiles.php?group_id=120651">Download</a>
<li><a href="http://sourceforge.net/tracker/?group_id=120651">Bug Tracker</a>
</ul>
</ul><a name="CHILD_LINKS"></a>
<ul class="ChildLinks">
<li><a href="front.html">Front Matter</a>
<li><a href="contents.html">Contents</a>
<ul>
<li><a href="node3.html">1 What is ALSA</a>
<li><a href="node4.html">2 ALSA and Python</a>
<li><a href="node5.html">3 Installation</a>
<li><a href="module-alsaaudio.html">4 <tt class="module">alsaaudio</tt></a>
<ul>
<li><a href="node7.html">4.1 PCM Terminology and Concepts</a>
<li><a href="pcm-objects.html">4.2 PCM Objects</a>
<li><a href="mixer-objects.html">4.3 Mixer Objects</a>
<li><a href="pcm-example.html">4.4 ALSA Examples</a>
</ul>
</ul>
<li><a href="about.html">About this document ...</a>
</ul>
<!--End of Table of Child-Links-->
</div>
<div class="navigation">
<div class='online-navigation'>
<p></p><hr />
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></td>
<td class='online-navigation'><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></td>
<td class='online-navigation'><a rel="next" title="Front Matter"
href="front.html"><img src='next.png'
border='0' height='32' alt='Next Page' width='32' /></a></td>
<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
<b class="navlabel">Next:</b>
<a class="sectref" rel="next" href="front.html">Front Matter</a>
</div>
</div>
<hr />
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->
</body>
</html>

BIN
doc/index.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 529 B

25
doc/index.rst Normal file
View File

@@ -0,0 +1,25 @@
alsaaudio documentation
=======================
.. toctree::
:maxdepth: 2
pyalsaaudio
libalsaaudio
SourceForge pages
=================
* `Project page <http://sourceforge.net/projects/pyalsaaudio/>`_
* `Download <http://sourceforge.net/project/showfiles.php?group_id=120651">`_
* `Bug tracker <http://sourceforge.net/tracker/?group_id=120651>`_
Indices and tables
==================
* :ref:`genindex`
* :ref:`modindex`
* :ref:`search`

533
doc/libalsaaudio.rst Normal file
View File

@@ -0,0 +1,533 @@
****************************
PCM Terminology and Concepts
****************************
In order to use PCM devices it is useful to be familiar with some concepts and
terminology.
Sample
PCM audio, whether it is input or output, consists of *samples*.
A single sample represents the amplitude of one channel of sound
at a certain point in time. A lot of individual samples are
necessary to represent actual sound; for CD audio, 44100 samples
are taken every second.
Samples can be of many different sizes, ranging from 8 bit to 64
bit precision. The specific format of each sample can also vary -
they can be big endian byte integers, little endian byte integers, or
floating point numbers.
Musically, the sample size determines the dynamic range. The
dynamic range is the difference between the quietest and the
loudest signal that can be resproduced.
Frame
A frame consists of exactly one sample per channel. If there is only one
channel (Mono sound) a frame is simply a single sample. If the sound is
stereo, each frame consists of two samples, etc.
Frame size
This is the size in bytes of each frame. This can vary a lot: if each sample
is 8 bits, and we're handling mono sound, the frame size is one byte.
Similarly in 6 channel audio with 64 bit floating point samples, the frame
size is 48 bytes
Rate
PCM sound consists of a flow of sound frames. The sound rate controls how
often the current frame is replaced. For example, a rate of 8000 Hz
means that a new frame is played or captured 8000 times per second.
Data rate
This is the number of bytes, which must be recorded or provided per
second at a certain frame size and rate.
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
8000 \* 1 \* 1 = 8 kb/s or 64kbit/s. This is typically used for telephony.
At the other end of the scale, 96000 Hz, 6 channel sound with 64
bit (8 bytes) samples has a data rate of 96000 \* 6 \* 8 = 4608
kb/s (almost 5 Mb sound data per second)
Period
When the hardware processes data this is done in chunks of frames. The time
interval between each processing (A/D or D/A conversion) is known
as the period.
The size of the period has direct implication on the latency of the
sound input or output. For low-latency the period size should be
very small, while low CPU resource usage would usually demand
larger period sizes. With ALSA, the CPU utilization is not impacted
much by the period size, since the kernel layer buffers multiple
periods internally, so each period generates an interrupt and a
memory copy, but userspace can be slower and read or write multiple
periods at the same time.
Period size
This is the size of each period in Hz. *Not bytes, but Hz!.* In
:mod:`alsaaudio` the period size is set directly, and it is
therefore important to understand the significance of this
number. If the period size is configured to for example 32,
each write should contain exactly 32 frames of sound data, and each
read will return either 32 frames of data or nothing at all.
Once you understand these concepts, you will be ready to use the PCM API. Read
on.
********************
Module documentation
********************
:mod:`alsaaudio`
================
.. module:: alsaaudio
:platform: Linux
.. % \declaremodule{builtin}{alsaaudio} % standard library, in C
.. % not standard, in C
.. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
.. moduleauthor:: Lars Immisch <lars@ibp.de>
.. % Author of the module code;
The :mod:`alsaaudio` module defines functions and classes for using ALSA.
.. % ---- 3.1. ----
.. % For each function, use a ``funcdesc'' block. This has exactly two
.. % parameters (each parameters is contained in a set of curly braces):
.. % the first parameter is the function name (this automatically
.. % generates an index entry); the second parameter is the function's
.. % argument list. If there are no arguments, use an empty pair of
.. % curly braces. If there is more than one argument, separate the
.. % arguments with backslash-comma. Optional parts of the parameter
.. % list are contained in \optional{...} (this generates a set of square
.. % brackets around its parameter). Arguments are automatically set in
.. % italics in the parameter list. Each argument should be mentioned at
.. % least once in the description; each usage (even inside \code{...})
.. % should be enclosed in \var{...}.
.. function:: mixers([cardname])
List the available mixers. The optional *cardname* specifies which card
should be queried (this is only relevant if you have more than one sound
card). Omit to use the default sound card.
.. class:: PCM([type], [mode], [cardname])
This class is used to represent a PCM device (both playback and capture
devices). The arguments are: --- *type* - can be either PCM_CAPTURE or
PCM_PLAYBACK (default). --- *mode* - can be either PCM_NONBLOCK,
PCM_ASYNC, or PCM_NORMAL (the default). --- *cardname* - 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
.. class:: Mixer([control], [id], [cardname])
This class is used to access a specific ALSA mixer. The arguments are: ---
*control* - Name of the chosen mixed (default is Master). --- *id* - id of
mixer (default is 0) -- More explanation needed here --- *cardname*
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.
.. exception:: ALSAAudioError
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:
PCM Objects
-----------
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.
PCM objects in :mod:`alsaaudio` 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:
.. class:: PCM([type], [mode], [cardname])
*type* - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
*mode* - 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.
In the current version of :mod:`alsaaudio` PCM_ASYNC is useless,
since it relies on a callback procedure, which can't be specified
through this API yet.
*cardname* - 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.
This will construct a PCM object with default settings:
Sample format: PCM_FORMAT_S16_LE --- Rate: 8000 Hz --- Channels: 2 ---
Period size: 32 frames ---
PCM objects have the following methods:
.. method:: PCM.pcmtype()
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
.. method:: PCM.pcmmode()
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC,
or PCM_NORMAL
.. method:: PCM.cardname()
Return the name of the sound card used by this PCM object.
.. method:: PCM.setchannels(nchannels)
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
.. method:: PCM.setrate(rate)
Set the sample rate in Hz for the device. Typical values are 8000
(mainly used for telephony), 16000, 44100 (CD quality), and 96000.
.. method:: PCM.setformat(format)
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
===================== ===============
.. method:: PCM.setperiodsize(period)
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)
.. method:: 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.
.. method:: 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.
.. method:: PCM.pause([enable=1])
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 the
people don't properly understand that writes to PCM devices must match
*exactly* 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 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.
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
acummulate 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
-------------
Mixer objects provides access to the ALSA mixer API.
.. class:: Mixer([control], [id], [cardname])
*control* - specifies which control to manipulate using this mixer
object. The list of available controls can be found with the
:mod:`alsaaudio`.\ :func:`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
*cardname* - 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.
Mixer objects have the following methods:
.. method:: Mixer.cardname()
Return the name of the sound card used by this Mixer object
.. method:: Mixer.mixer()
Return the name of the specific mixer controlled by this object, For example
'Master' or 'PCM'
.. method:: Mixer.mixerid()
Return the ID of the ALSA mixer controlled by this object.
.. method:: 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 swithes use the :meth:`setrec` or
:meth:`setmute` methods
.. method:: 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
====================== ================
.. method:: 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 :mod:`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.
.. method:: 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.
.. method:: Mixer.getrange([direction])
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'
.. method:: 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.
.. method:: Mixer.getvolume([direction])
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'
.. method:: Mixer.setvolume(volume,[channel], [direction])
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'.
.. method:: 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
.. method:: 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.
**A Note on the ALSA Mixer API**
The ALSA mixer API is extremely complicated - and hardly documented at all.
:mod:`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 :mod:`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.
.. % ==== 4. ====
.. _pcm-example:
ALSA Examples
-------------
For now, the only examples available are the 'playbacktest.py' and the
'recordtest.py' programs included. This will change in a future version.

BIN
doc/modules.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 385 B

BIN
doc/modules.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 598 B

BIN
doc/next.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 253 B

BIN
doc/next.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 511 B

BIN
doc/previous.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 252 B

BIN
doc/previous.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 511 B

243
doc/pyalsaaudio.css
View File

@@ -1,243 +0,0 @@
/*
* The first part of this is the standard CSS generated by LaTeX2HTML,
* with the "empty" declarations removed.
*/
/* Century Schoolbook font is very similar to Computer Modern Math: cmmi */
.math { font-family: "Century Schoolbook", serif; }
.math i { font-family: "Century Schoolbook", serif;
font-weight: bold }
.boldmath { font-family: "Century Schoolbook", serif;
font-weight: bold }
/*
* Implement both fixed-size and relative sizes.
*
* I think these can be safely removed, as it doesn't appear that
* LaTeX2HTML ever generates these, even though these are carried
* over from the LaTeX2HTML stylesheet.
*/
small.xtiny { font-size : xx-small; }
small.tiny { font-size : x-small; }
small.scriptsize { font-size : smaller; }
small.footnotesize { font-size : small; }
big.xlarge { font-size : large; }
big.xxlarge { font-size : x-large; }
big.huge { font-size : larger; }
big.xhuge { font-size : xx-large; }
/*
* Document-specific styles come next;
* these are added for the Python documentation.
*
* Note that the size specifications for the H* elements are because
* Netscape on Solaris otherwise doesn't get it right; they all end up
* the normal text size.
*/
body { color: #000000;
background-color: #ffffff; }
a:link:active { color: #ff0000; }
a:link:hover { background-color: #bbeeff; }
a:visited:hover { background-color: #bbeeff; }
a:visited { color: #551a8b; }
a:link { color: #0000bb; }
h1, h2, h3, h4, h5, h6 { font-family: avantgarde, sans-serif;
font-weight: bold; }
h1 { font-size: 180%; }
h2 { font-size: 150%; }
h3, h4 { font-size: 120%; }
/* These are section titles used in navigation links, so make sure we
* match the section header font here, even it not the weight.
*/
.sectref { font-family: avantgarde, sans-serif; }
/* And the label before the titles in navigation: */
.navlabel { font-size: 85%; }
/* LaTeX2HTML insists on inserting <br> elements into headers which
* are marked with \label. This little bit of CSS magic ensures that
* these elements don't cause spurious whitespace to be added.
*/
h1>br, h2>br, h3>br,
h4>br, h5>br, h6>br { display: none; }
code, tt { font-family: "lucida typewriter", lucidatypewriter,
monospace; }
var { font-family: times, serif;
font-style: italic;
font-weight: normal; }
.Unix { font-variant: small-caps; }
.typelabel { font-family: lucida, sans-serif; }
.navigation td { background-color: #99ccff;
font-weight: bold;
font-family: avantgarde, sans-serif;
font-size: 110%; }
div.warning { background-color: #fffaf0;
border: thin solid black;
padding: 1em;
margin-left: 2em;
margin-right: 2em; }
div.warning .label { font-family: sans-serif;
font-size: 110%;
margin-right: 0.5em; }
div.note { background-color: #fffaf0;
border: thin solid black;
padding: 1em;
margin-left: 2em;
margin-right: 2em; }
div.note .label { margin-right: 0.5em;
font-family: sans-serif; }
address { font-size: 80%; }
.release-info { font-style: italic;
font-size: 80%; }
.titlegraphic { vertical-align: top; }
.verbatim pre { color: #00008b;
font-family: "lucida typewriter", lucidatypewriter,
monospace;
font-size: 90%; }
.verbatim { margin-left: 2em; }
.verbatim .footer { padding: 0.05in;
font-size: 85%;
background-color: #99ccff;
margin-right: 0.5in; }
.grammar { background-color: #99ccff;
margin-right: 0.5in;
padding: 0.05in; }
.grammar-footer { padding: 0.05in;
font-size: 85%; }
.grammartoken { font-family: "lucida typewriter", lucidatypewriter,
monospace; }
.productions { background-color: #bbeeff; }
.productions a:active { color: #ff0000; }
.productions a:link:hover { background-color: #99ccff; }
.productions a:visited:hover { background-color: #99ccff; }
.productions a:visited { color: #551a8b; }
.productions a:link { color: #0000bb; }
.productions table { vertical-align: baseline;
empty-cells: show; }
.productions > table td,
.productions > table th { padding: 2px; }
.productions > table td:first-child,
.productions > table td:last-child {
font-family: "lucida typewriter",
lucidatypewriter,
monospace;
}
/* same as the second selector above, but expressed differently for Opera */
.productions > table td:first-child + td + td {
font-family: "lucida typewriter",
lucidatypewriter,
monospace;
vertical-align: baseline;
}
.productions > table td:first-child + td {
padding-left: 1em;
padding-right: 1em;
}
.productions > table tr { vertical-align: baseline; }
.email { font-family: avantgarde, sans-serif; }
.mailheader { font-family: avantgarde, sans-serif; }
.mimetype { font-family: avantgarde, sans-serif; }
.newsgroup { font-family: avantgarde, sans-serif; }
.url { font-family: avantgarde, sans-serif; }
.file { font-family: avantgarde, sans-serif; }
.guilabel { font-family: avantgarde, sans-serif; }
.realtable { border-collapse: collapse;
border-color: black;
border-style: solid;
border-width: 0px 0px 2px 0px;
empty-cells: show;
margin-left: auto;
margin-right: auto;
padding-left: 0.4em;
padding-right: 0.4em;
}
.realtable tbody { vertical-align: baseline; }
.realtable tfoot { display: table-footer-group; }
.realtable thead { background-color: #99ccff;
border-width: 0px 0px 2px 1px;
display: table-header-group;
font-family: avantgarde, sans-serif;
font-weight: bold;
vertical-align: baseline;
}
.realtable thead :first-child {
border-width: 0px 0px 2px 0px;
}
.realtable thead th { border-width: 0px 0px 2px 1px }
.realtable td,
.realtable th { border-color: black;
border-style: solid;
border-width: 0px 0px 1px 1px;
padding-left: 0.4em;
padding-right: 0.4em;
}
.realtable td:first-child,
.realtable th:first-child {
border-left-width: 0px;
vertical-align: baseline;
}
.center { text-align: center; }
.left { text-align: left; }
.right { text-align: right; }
.refcount-info { font-style: italic; }
.refcount-info .value { font-weight: bold;
color: #006600; }
/*
* Some decoration for the "See also:" blocks, in part inspired by some of
* the styling on Lars Marius Garshol's XSA pages.
* (The blue in the navigation bars is #99CCFF.)
*/
.seealso { background-color: #fffaf0;
border: thin solid black;
padding: 0pt 1em 4pt 1em; }
.seealso > .heading { font-size: 110%;
font-weight: bold; }
/*
* Class 'availability' is used for module availability statements at
* the top of modules.
*/
.availability .platform { font-weight: bold; }
/*
* Additional styles for the distutils package.
*/
.du-command { font-family: monospace; }
.du-option { font-family: avantgarde, sans-serif; }
.du-filevar { font-family: avantgarde, sans-serif;
font-style: italic; }
.du-xxx:before { content: "** ";
font-weight: bold; }
.du-xxx:after { content: " **";
font-weight: bold; }
/*
* Some specialization for printed output.
*/
@media print {
.online-navigation { display: none; }
}

107
doc/pyalsaaudio.rst Normal file
View File

@@ -0,0 +1,107 @@
***************
PyAlsaAudio
***************
:Author: Casper Wilstrup
:Author: Lars Immisch
.. |release| replace:: 0.4
.. % At minimum, give your name and an email address. You can include a
.. % snail-mail address if you like.
.. % This makes the Abstract go on a separate page in the HTML version;
.. % if a copyright notice is used, it should go immediately after this.
.. %
.. _front:
This software is licensed under the PSF license - the same one used by the
majority of the python distribution. Basically you can use it for anything you
wish (even commercial purposes). There is no warranty whatsoever.
.. % Copyright statement should go here, if needed.
.. % The abstract should be a paragraph or two long, and describe the
.. % scope of the document.
.. topic:: Abstract
This package contains wrappers for accessing the ALSA API from Python. It is
currently fairly complete for PCM devices and Mixer access. MIDI sequencer
support is low on our priority list, but volunteers are welcome.
If you find bugs in the wrappers please use the SourceForge bug tracker.
Please don't send bug reports regarding ALSA specifically. There are several
bugs in this API, and those should be reported to the ALSA team - not me.
************
What is ALSA
************
The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI
functionality to the Linux operating system.
Logically ALSA consists of these components:
* A set of kernel drivers. --- These drivers are responsible for handling the
physical sound hardware from within the Linux kernel, and have been the
standard sound implementation in Linux since kernel version 2.5
* A kernel level API for manipulating the ALSA devices.
* A user-space C library for simplified access to the sound hardware from
userspace applications. This library is called *libasound* and is required by
all ALSA capable applications.
More information about ALSA may be found on the project homepage
`<http://www.alsa-project.org>`_
ALSA and Python
===============
The older Linux sound API (OSS) which is now deprecated is well supported from
the standard Python library, through the ossaudiodev module. No native ALSA
support exists in the standard library.
There are a few other "ALSA for Python" projects available, including at least
two different projects called pyAlsa. Neither of these seem to be under active
development at the time - and neither are very feature complete.
I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module
included in the standard Python library, but that looks currently unlikely.
PyAlsaAudio hass full support for sound capture, playback of sound, as well as
the ALSA Mixer API.
MIDI support is not available, and since I don't own any MIDI hardware, it's
difficult for me to implement it. Volunteers to work on this would be greatly
appreciated.
************
Installation
************
Note: the wrappers link with the alsasound library (from the alsa-lib package)
and need the ALSA headers for compilation. Verify that you have
/usr/lib/libasound.so and /usr/include/alsa (or similar paths) before building.
On Debian (and probably Ubuntu), make sure you have libasound2-dev installed.
Naturally you also need to use a kernel with proper ALSA support. This is the
default in Linux kernel 2.6 and later. If you are using kernel version 2.4 you
may need to install the ALSA patches yourself - although most distributions
ship with ALSA kernels.
To install, execute the following: --- ::
$ python setup.py build
And then as root: --- ::
# python setup.py install

BIN
doc/pyfav.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 125 B

BIN
doc/pyfav.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 240 B

23
doc/src/Makefile
View File

@@ -1,23 +0,0 @@
# Makefile for PyAlsaAudio documentation
#
# In order to use this you need an unpacked version of the
# Python source available. Set the source path below
#
# You also need a working latex installation, and the latex2html
# tool installed.
PYTHONSOURCE = /usr/local/src/Python-2.5.1/
SFUSER = larsimmisch
# Shouldn't need to change anything below here!
MKHOWTO = $(PYTHONSOURCE)/Doc/tools/mkhowto
all:
$(MKHOWTO) --dir .. --html pyalsaaudio.tex
python sffixup.py ../index.html
text:
$(MKHOWTO) --dir .. --text pyalsaaudio.tex
install:
scp ../*.html ../*.gif ../*.png $(SFUSER)@shell.sourceforge.net:/home/groups/p/py/pyalsaaudio/htdocs

521
doc/src/libalsaaudio.tex
View File

@@ -1,521 +0,0 @@
\section{\module{alsaaudio}}
%\declaremodule{builtin}{alsaaudio} % standard library, in C
\declaremodule{extension}{alsaaudio} % not standard, in C
\platform{Linux}
\moduleauthor{Casper Wilstrup}{cwi@aves.dk} % Author of the module code;
\modulesynopsis{ALSA sound support}
The \module{alsaaudio} module defines functions and classes for using
ALSA.
% ---- 3.1. ----
% For each function, use a ``funcdesc'' block. This has exactly two
% parameters (each parameters is contained in a set of curly braces):
% the first parameter is the function name (this automatically
% generates an index entry); the second parameter is the function's
% argument list. If there are no arguments, use an empty pair of
% curly braces. If there is more than one argument, separate the
% arguments with backslash-comma. Optional parts of the parameter
% list are contained in \optional{...} (this generates a set of square
% brackets around its parameter). Arguments are automatically set in
% italics in the parameter list. Each argument should be mentioned at
% least once in the description; each usage (even inside \code{...})
% should be enclosed in \var{...}.
\begin{funcdesc}{mixers}{\optional{cardname}}
List the available mixers. The optional \var{cardname} specifies which
card should be queried (this is only relevant if you have more than one
sound card). Omit to use the default sound card.
\end{funcdesc}
\begin{classdesc}{PCM}{\optional{type}, \optional{mode}, \optional{cardname}}
This class is used to represent a PCM device (both playback and
capture devices).
The arguments are: \\
\var{type} - can be either PCM_CAPTURE or PCM_PLAYBACK (default). \\
\var{mode} - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).\\
\var{cardname} - 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
\end{classdesc}
\begin{classdesc}{Mixer}{\optional{control}, \optional{id}, \optional{cardname}}
This class is used to access a specific ALSA mixer.
The arguments are: \\
\var{control} - Name of the chosen mixed (default is Master). \\
\var{id} - id of mixer (default is 0) -- More explanation needed here\\
\var{cardname} 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
\end{classdesc}
\begin{excdesc}{ALSAAudioError}
Exception raised when an operation fails for a ALSA specific reason.
The exception argument is a string describing the reason of the
failure.
\end{excdesc}
\subsection{PCM Terminology and Concepts}
In order to use PCM devices it is useful to be familiar with some concepts and
terminology.
\begin{description}
\item[Sample] PCM audio, whether it is input or output, consists at
the lowest level of a number of single samples. A sample represents
the sound in a single channel in a brief interval. If more than one
channel is in use, more than one sample is required for each
interval to describe the sound. Samples can be of many different
sizes, ranging from 8 bit to 64 bit presition. The specific format
of each sample can also vary - they can be big endian byte order,
little endian byte order, or even floats.
\item[Frame] A frame consists of exactly one sample per channel. If
there is only one channel (Mono sound) a frame is simply a single
sample. If the sound is stereo, each frame consists of two samples,
etc.
\item[Frame size] This is the size in bytes of each frame. This can
vary a lot: if each sample is 8 bits, and we're handling mono sound,
the frame size is one byte. Similarly in 6 channel audio with 64 bit
floating point samples, the frame size is 48 bytes
\item[Rate] PCM sound consists of a flow of sound frames. The sound
rate controls how often the current frame is replaced. For example,
a rate of 8000 Hz means that a new frame is played or captured 8000
times per second.
\item[Data rate] This is the number of bytes, which must be recorded
or provided per second at a certain frame size and rate.
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
8000 * 1 * 1 = 8 kb/s
At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit
(8 bytes) samples has a data rate of 96000 * 6 * 8 = 4608 kb/s
(almost 5 Mb sound data per second)
\item[Period] When the hardware processes data this is done in chunks
of frames. The time interval between each processing (A/D or D/A
conversion) is known as the period. The size of the period has
direct implication on the latency of the sound input or output. For
low-latency the period size should be very small, while low CPU
resource usage would usually demand larger period sizes. With ALSA,
the CPU utilization is not impacted much by the period size, since
the kernel layer buffers multiple periods internally, so each period
generates an interrupt and a memory copy, but userspace can be
slower and read or write multiple periods at the same time.
\item[Period size] This is the size of each period in Hz. \emph{Not
bytes, but Hz!.} In \module{alsaaudio} the period size is set
directly, and it is therefore important to understand the
significance of this number. If the period size is configured to for
example 32, each write should contain exactly 32 frames of sound
data, and each read will return either 32 frames of data or nothing
at all.
\end{description}
Once you understand these concepts, you will be ready to use the PCM
API. Read on.
\subsection{PCM Objects}
\label{pcm-objects}
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.
PCM objects in \module{alsaaudio} 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:
\begin{classdesc}{PCM}{\optional{type}, \optional{mode}, \optional{cardname}}
\var{type} - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
\var{mode} - 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.
In the current version of \module{alsaaudio} PCM_ASYNC is useless,
since it relies on a callback procedure, which can't be specified through
this API yet.
\var{cardname} - 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
This will construct a PCM object with default settings:
Sample format: PCM_FORMAT_S16_LE \\
Rate: 8000 Hz \\
Channels: 2 \\
Period size: 32 frames \\
\end{classdesc}
PCM objects have the following methods:
\begin{methoddesc}[PCM]{pcmtype}{}
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
\end{methoddesc}
\begin{methoddesc}[PCM]{pcmmode}{}
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC,
or PCM_NORMAL
\end{methoddesc}
\begin{methoddesc}[PCM]{cardname}{}
Return the name of the sound card used by this PCM object.
\end{methoddesc}
\begin{methoddesc}[PCM]{setchannels}{nchannels}
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
\end{methoddesc}
\begin{methoddesc}[PCM]{setrate}{rate}
Set the sample rate in Hz for the device. Typical values are 8000
(poor sound), 16000, 44100 (cd quality), and 96000
\end{methoddesc}
\begin{methoddesc}[PCM]{setformat}{format}
The sound \var{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:
\begin{tableii}{l|l}{Formats}{Format}{Description}
\lineii{PCM_FORMAT_S8}{Signed 8 bit samples for each channel}
\lineii{PCM_FORMAT_U8}{Signed 8 bit samples for each channel}
\lineii{PCM_FORMAT_S16_LE}{Signed 16 bit samples for each channel
(Little Endian byte order)}
\lineii{PCM_FORMAT_S16_BE}{Signed 16
bit samples for each channel (Big Endian byte order)}
\lineii{PCM_FORMAT_U16_LE}{Unsigned 16 bit samples for each channel
(Little Endian byte order)}
\lineii{PCM_FORMAT_U16_BE}{Unsigned 16
bit samples for each channel (Big Endian byte order)}
\lineii{PCM_FORMAT_S24_LE}{Signed 24 bit samples for each channel
(Little Endian byte order)}
\lineii{PCM_FORMAT_S24_BE}{Signed 24
bit samples for each channel (Big Endian byte order)}
\lineii{PCM_FORMAT_U24_LE}{Unsigned 24 bit samples for each channel
(Little Endian byte order)}
\lineii{PCM_FORMAT_U24_BE}{Unsigned 24
bit samples for each channel (Big Endian byte order)}
\lineii{PCM_FORMAT_S32_LE}{Signed 32 bit samples for each channel
(Little Endian byte order)}
\lineii{PCM_FORMAT_S32_BE}{Signed 32
bit samples for each channel (Big Endian byte order)}
\lineii{PCM_FORMAT_U32_LE}{Unsigned 32 bit samples for each channel
(Little Endian byte order)}
\lineii{PCM_FORMAT_U32_BE}{Unsigned 32
bit samples for each channel (Big Endian byte order)}
\lineii{PCM_FORMAT_FLOAT_LE}{32 bit samples encoded as float.
(Little Endian byte order)}
\lineii{PCM_FORMAT_FLOAT_BE}{32 bit
samples encoded as float (Big Endian byte order)}
\lineii{PCM_FORMAT_FLOAT64_LE}{64 bit samples encoded as float.
(Little Endian byte order)}
\lineii{PCM_FORMAT_FLOAT64_BE}{64 bit
samples encoded as float. (Big Endian byte order)}
\lineii{PCM_FORMAT_MU_LAW}{A logarithmic encoding (used by Sun .au
files)}
\lineii{PCM_FORMAT_A_LAW}{Another logarithmic encoding}
\lineii{PCM_FORMAT_IMA_ADPCM}{a 4:1 compressed format defined by the
Interactive Multimedia Association} \lineii{PCM_FORMAT_MPEG}{MPEG
encoded audio?}
\lineii{PCM_FORMAT_GSM}{9600 bits/s constant rate encoding for speech}
\end{tableii}
\end{methoddesc}
\begin{methoddesc}[PCM]{setperiodsize}{period}
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)
\end{methoddesc}
\begin{methoddesc}[PCM]{read}{}
In PCM_NORMAL mode, this function blocks until a full period is
available, and then returns a tuple (length,data) where
\emph{length} is the number of frames of captured data, and
\emph{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
\code{(0,'')} if no new period has become available since the last
call to read.
\end{methoddesc}
\begin{methoddesc}[PCM]{write}{data}
Writes (plays) the sound in data. The length of data \emph{must} be
a multiple of the frame size, and \emph{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.
\end{methoddesc}
\begin{methoddesc}[PCM]{pause}{\optional{enable=1}}
If \var{enable} is 1, playback or capture is paused. If \var{enable} is 0,
playback/capture is resumed.
\end{methoddesc}
\strong{A few hints on using PCM devices for playback}
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 \emph{exactly} 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 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.
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
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.
\subsection{Mixer Objects}
\label{mixer-objects}
Mixer objects provides access to the ALSA mixer API.
\begin{classdesc}{Mixer}{\optional{control}, \optional{id},
\optional{cardname}}
\var{control} - specifies which control to manipulate using this
mixer object. The list of available controls can be found with the
\module{alsaaudio}.\function{mixers} function. The default value is
'Master' - other common controls include 'Master Mono', 'PCM',
'Line', etc.
\var{id} - the id of the mixer control. Default is 0
\var{cardname} - 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
\end{classdesc}
Mixer objects have the following methods:
\begin{methoddesc}[Mixer]{cardname}{}
Return the name of the sound card used by this Mixer object
\end{methoddesc}
\begin{methoddesc}[Mixer]{mixer}{}
Return the name of the specific mixer controlled by this object, For
example 'Master' or 'PCM'
\end{methoddesc}
\begin{methoddesc}[Mixer]{mixerid}{}
Return the ID of the ALSA mixer controlled by this object.
\end{methoddesc}
\begin{methoddesc}[Mixer]{switchcap}{}
Returns a list of the switches which are defined by this specific
mixer. Possible values in this list are:
\begin{tableii}{l|l}{Switches}{Switch}{Description}
\lineii{'Mute'}{This mixer can be muted}
\lineii{'Joined Mute'}{This mixer can mute all channels at the same time}
\lineii{'Playback Mute'}{This mixer can mute the playback output}
\lineii{'Joined Playback Mute'}
{Mute playback for all channels at the same time}
\lineii{'Capture Mute'}{Mute sound capture}
\lineii{'Joined Capture Mute'}{Mute sound capture for all channels at a time}
\lineii{'Capture Exclusive'}{Not quite sure what this is}
\end{tableii}
To manipulate these swithes use the \method{setrec} or
\method{setmute} methods
\end{methoddesc}
\begin{methoddesc}[Mixer]{volumecap}{}
Returns a list of the volume control capabilities of this mixer.
Possible values in the list are:
\begin{tableii}{l|l}{Volume Capabilities}{Capability}{Description}
\lineii{'Volume'}{This mixer can control volume}
\lineii{'Joined Volume'}{This mixer can control volume for all channels at
the same time}
\lineii{'Playback Volume'}{This mixer can manipulate the playback volume}
\lineii{'Joined Playback Volume'}{Manipulate playback volumne for all
channels at the same time}
\lineii{'Capture Volume'}{Manipulate sound capture volume}
\lineii{'Joined Capture Volume'}{Manipulate sound capture volume for all
channels at a time}
\end{tableii}
\end{methoddesc}
\begin{methoddesc}[Mixer]{getenum}{}
For enumerated controls, return the currently selected item and
the list of items available.
Returns a tuple \textit{(string, list of strings)}.
For example, my soundcard has a Mixer called \textit{Mono Output Select}.
Using \textit{amixer}, I get:
\begin{verbatim}
$ amixer get "Mono Output Select"
Simple mixer control 'Mono Output Select',0
Capabilities: enum
Items: 'Mix' 'Mic'
Item0: 'Mix'
\end{verbatim}
Using \module{alsaaudio}, one could do:
\begin{verbatim}
>>> import alsaaudio
>>> m = alsaaudio.Mixer('Mono Output Select')
>>> m.getenum()
('Mix', ['Mix', 'Mic'])
\end{verbatim}
This method will return an empty tuple if the mixer is not an
enumerated control.
\end{methoddesc}
\begin{methoddesc}[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.
\end{methoddesc}
\begin{methoddesc}[Mixer]{getrange}{\optional{direction}}
Return the volume range of the ALSA mixer controlled by this object.
The optional \var{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'
\end{methoddesc}
\begin{methoddesc}[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.
\end{methoddesc}
\begin{methoddesc}[Mixer]{getvolume}{\optional{direction}}
Returns a list with the current volume settings for each channel.
The list elements are integer percentages.
The optional \var{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'
\end{methoddesc}
\begin{methoddesc}[Mixer]{setvolume}{volume,\optional{channel},
\optional{direction}}
Change the current volume settings for this mixer. The \var{volume}
argument controls the new volume setting as an integer percentage.
If the optional argument \var{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 \var{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'.
\end{methoddesc}
\begin{methoddesc}[Mixer]{setmute}{mute, \optional{channel}}
Sets the mute flag to a new value. The \var{mute} argument is either
0 for not muted, or 1 for muted.
The optional \var{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
\end{methoddesc}
\begin{methoddesc}[Mixer]{setrec}{capture,\optional{channel}}
Sets the capture mute flag to a new value. The \var{capture}
argument is either 0 for no capture, or 1 for capture.
The optional \var{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.
\end{methoddesc}
\textbf{A Note on the ALSA Mixer API}
The ALSA mixer API is extremely complicated - and hardly documented at
all. \module{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 \module{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 need into future versions.
If the current state of affairs annoy 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.
% ==== 4. ====
\subsection{ALSA Examples \label{pcm-example}}
For now, the only examples available are the 'playbacktest.py' and the
'recordtest.py' programs included. This will change in a future
version.

115
doc/src/pyalsaaudio.tex
View File

@@ -1,115 +0,0 @@
\documentclass{howto}
\title{PyAlsaAudio}
\release{0.4}
% At minimum, give your name and an email address. You can include a
% snail-mail address if you like.
\author{Casper Wilstrup}
\authoraddress{cwi@aves.dk}
\begin{document}
\maketitle
% This makes the Abstract go on a separate page in the HTML version;
% if a copyright notice is used, it should go immediately after this.
%
\ifhtml
\chapter*{Front Matter\label{front}}
\fi
% Copyright statement should go here, if needed.
This software is licensed under the PSF license - the same one used
by the majority of the python distribution. Basically you can use it
for anything you wish (even commercial purposes). There is no warranty
whatsoever.
% The abstract should be a paragraph or two long, and describe the
% scope of the document.
\begin{abstract}
\noindent
This package contains wrappers for accessing the ALSA API from Python.
It is currently fairly complete for PCM devices and Mixer access. MIDI
sequencer support is low on my priority list, but volunteers are
welcome.
If you find bugs in the wrappers please use the SourceForge bug
tracker. Please don't send bug reports regarding ALSA specifically.
There are several bugs in this API, and those should be reported to
the ALSA team - not me.
\end{abstract}
\tableofcontents
\section{What is ALSA}
The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI
functionality to the Linux operating system.
Logically ALSA consists of these components:
\begin{itemize}
\item A set of kernel drivers.\\
These drivers are responsible for handling the physical sound
hardware from within the Linux kernel, and have been the standard
sound implementation in Linux since kernel version 2.5
\item A kernel level API for manipulating the ALSA devices.
\item A user-space C library for simplified access to the sound hardware
from userspace applications. This library is called \textit{libasound}
and is required by all ALSA capable applications.
\end{itemize}
More information about ALSA may be found on the project homepage
\url{http://www.alsa-project.org}
\section{ALSA and Python}
The older Linux sound API (OSS) which is now deprecated is well
supported from the standard Python library, through the ossaudiodev
module. No native ALSA support exists in the standard library (yet).
There are a few other ``ALSA for Python'' projects available,
including at least two different projects called pyAlsa. Neither of
these seem to be under active development at the time - and neither
are very feature complete.
I wrote PyAlsaAudio to fill this gap. My long term goal is to have the
module included in the standard Python library, but that is probably a
while off yet.
PyAlsaAudio hass full support for sound capture, playback of sound, as
well as the ALSA Mixer API.
MIDI support is not available, and since I don't own any MIDI
hardware, it's difficult for me to implement it. Volunteers to work on
this would be greatly appreciated
\section{Installation}
Note: the wrappers link with the alsasound library (from the alsa-lib
package) and need the ALSA headers for compilation. Verify that you
have /usr/lib/libasound.so and /usr/include/alsa (or
similar paths) before building.
On Debian (and probably Ubuntu), make sure you have libasound2-dev installed.
Naturally you also need to use a kernel with proper ALSA support. This
is the default in Linux kernel 2.6 and later. If you are using kernel
version 2.4 you may need to install the ALSA patches yourself -
although most distributions ship with ALSA kernels.
To install, execute the following: \\
\begin{verbatim}
$ python setup.py build
\end{verbatim}
And then as root: \\
\begin{verbatim}
# python setup.py install
\end{verbatim}
\input{libalsaaudio}
\end{document}

35
doc/src/sffixup.py
View File

@@ -1,35 +0,0 @@
#!/usr/bin/env/python
# This is a hack and totally brittle. Let's not talk about it.
import sys
sfnotes = '''<ul>
<li><a href="http://sourceforge.net/">SourceForge</li>
<ul>
<li><a href="http://sourceforge.net/projects/pyalsaaudio/">Summary page</a>
<li><a href="http://sourceforge.net/project/showfiles.php?group_id=120651">Download</a>
<li><a href="http://sourceforge.net/tracker/?group_id=120651">Bug Tracker</a>
</ul>
</ul>'''
if len(sys.argv) <= 1:
print "usage: sffixup.py <file>"
sys.exit(2)
f = open(sys.argv[1], 'r')
lines = f.readlines()
f.close()
try:
index = lines.index('<a name="CHILD_LINKS"></a>\n')
except ValueError:
print "Cannot find child links. SourceForge links will not appear on index.html."
sys.exit(1)
lines.insert(index, sfnotes)
f = open(sys.argv[1], 'w')
for l in lines:
f.write(l)
f.close()

BIN
doc/up.gif
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 316 B

BIN
doc/up.png
View File

Binary file not shown.
Side by Side

Before

Width:  |  Height:  |  Size: 577 B

2
playbacktest.py
View File

@@ -1,4 +1,4 @@
## recordtest.py
## playbacktest.py
##
## This is an example of a simple sound playback script.
##

89
playwav.py Executable file
View File

@@ -0,0 +1,89 @@
#!/usr/bin/env python
# Simple test script that plays (some) wav files
import sys
import struct
import alsaaudio
# this is all a bit simplified, and won't cope with any wav extensions
# or multiple data chunks, but it's good enough here
WAV_FORMAT_PCM = 1
WAV_FORMAT_ALAW = 6
WAV_FORMAT_MULAW = 7
WAV_HEADER = '<4sl4s4slhhllhh4sl'
WAV_HEADER_SIZE = struct.calcsize(WAV_HEADER)
def _b(s):
'Helper for 3.0 compatibility'
if sys.version_info[0] >= 3:
return bytes(s, 'UTF-8')
return s
def wav_header_unpack(data):
(riff, riffsize, wave, fmt, fmtsize, format, nchannels, framerate,
datarate, blockalign, bitspersample, data, datalength) \
= struct.unpack(WAV_HEADER, data)
print(data)
if riff != _b('RIFF') or fmtsize != 16 or fmt != _b('fmt ') \
or data != _b('data'):
raise ValueError('wav header too complicated')
return (format, nchannels, framerate, bitspersample, datalength)
def play(device, f):
header = f.read(WAV_HEADER_SIZE)
format, nchannels, framerate, bitspersample, datalength \
= wav_header_unpack(header)
# Set attributes
device.setchannels(nchannels)
device.setrate(framerate)
# We assume signed data, little endian
if format == WAV_FORMAT_PCM:
if bitspersample == 8:
device.setformat(alsaaudio.PCM_FORMAT_S8)
elif bitspersample == 16:
device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
elif bitspersample == 24:
device.setformat(alsaaudio.PCM_FORMAT_S24_LE)
elif bitspersample == 32:
device.setformat(alsaaudio.PCM_FORMAT_S32_LE)
elif format == WAV_FORMAT_ALAW:
device.setformat(alsaaudio.PCM_FORMAT_A_LAW)
elif format == WAV_FORMAT_MULAW:
device.setformat(alsaaudio.PCM_FORMAT_MU_LAW)
else:
raise ValueError('Unsupported format %d' % format)
# The period size controls the internal number of frames per period.
# The significance of this parameter is documented in the ALSA api.
# rs = framerate / 25
# out.setperiodsize(rs)
data = f.read()
while data:
# Read data from stdin
device.write(data)
data = f.read()
if __name__ == '__main__':
if len(sys.argv) < 2:
print('usage: playwav.py <file>')
sys.exit(2)
f = open(sys.argv[1], 'rb')
device = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK)
play(device, f)
f.close()