Report volume like alsamixer. Maybe.

I have no idea how they calculate dB.

.gitignore

@@ -1,7 +1,9 @@
 *.pyc
 *.so
+MANIFEST
 doc/gh-pages/
 doc/html/
 doc/doctrees/
 gh-pages/
-build/
+build/
+dist/

CHANGES

@@ -1,9 +1,40 @@
+Version 0.8.2:
+- fix #3 (we cannot get the revision from git for pip installs)
+
+Version 0.8.1:
+- document changes (this file)
+
+Version 0.8:
+- 'PCM()' has new 'device' and 'cardindex' keyword arguments.
+
+  The keyword 'device' allows to select virtual devices, 'cardindex' can be
+  used to select hardware cards by index (as with 'mixers()' and 'Mixer()').
+
+  The 'card' keyword argument is still supported, but deprecated.
+
+  The reason for this change is that the 'card' keyword argument guessed
+  a device name from the card name, but this only works sometimes, and breaks
+  opening virtual devices.
+
+- new function 'pcms()' to list available PCM devices.
+
+- mixers() and Mixer() take an additional 'device' keyword argument.
+  This allows to list or open virtual devices.
+
+- The default behaviour of Mixer() without any arguments has changed.
+  Now Mixer() will try to open the 'default' Mixer instead of the Mixer
+  that is associated with card 0.
+
+- fix a memory leak under Python 3.x
+
+- some more memory leaks in error conditions fixed.
+
 Version 0.7:
 - fixed several memory leaks (patch 3372909), contributed by Erik Kulyk)
 
 
 Version 0.6:
-- mostly reverted patch 2594366: alsapcm_setup did not do complete error 
+- mostly reverted patch 2594366: alsapcm_setup did not do complete error
 checking for good reasons; some ALSA functions in alsapcm_setup may fail without
 rendering the device unusable
 
@@ -15,7 +46,7 @@ Version 0.5:
 
 
 Version 0.4:
-- API changes: mixers() and Mixer() now take a card index instead of a 
+- API changes: mixers() and Mixer() now take a card index instead of a
   card name as optional parameter.
 - Support for Python 3.0
 - Documentation converted to reStructuredText; use Sphinx instead of LaTeX.

NOTES.md

@@ -0,0 +1,11 @@
+# Publishing the documentation
+
+ - Install Sphinx; `sudo pip install sphinx`
+ - Clone gh-pages branch: `cd doc; git clone -b gh-pages git@github.com:larsimmisch/pyalsaaudio.git gh-pages`
+ - `cd doc; make publish`
+
+# Release procedure
+
+ - Update version number in setup.py
+ - Create tag and push it, i.e. `git tag x.y.z; git push origin x.y.z`
+ - `python setup.py sdist upload -r pypi` 

README.md

@@ -5,12 +5,13 @@ For documentation, see http://larsimmisch.github.io/pyalsaaudio/
 > Author: Casper Wilstrup (cwi@aves.dk)  
 > Maintainer: Lars Immisch (lars@ibp.de)
 
-This package contains wrappers for accessing the ALSA api from Python. It
+This package contains wrappers for accessing the
+[ALSA](http://www.alsa-project.org/) API from Python. It
 is currently fairly complete for PCM devices, and has some support for mixers.
 
 If you find bugs in the wrappers please open an issue in the issue 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
+bugs in the ALSA API, and those should be reported to the ALSA team - not
 me.
 
 This software is licensed under the PSF license - the same one used

alsaaudio.c

@@ -20,7 +20,13 @@
 #define PyLong_Check PyInt_Check
 #define PyLong_AS_LONG PyInt_AS_LONG
 #endif
+
+#if PY_MAJOR_VERSION < 3
+    #define PyLong_FromLong PyInt_FromLong
+#endif
+
 #include <alsa/asoundlib.h>
+#include <alsa/version.h>
 #include <stdio.h>
 
 PyDoc_STRVAR(alsaaudio_module_doc,
@@ -71,6 +77,12 @@ typedef struct {
     snd_mixer_t *handle;
 } alsamixer_t;
 
+typedef enum {
+    unit_percent,
+    unit_dB,
+    unit_last
+} volume_unit_t;   
+
 /******************************************/
 /* PCM object wrapper                   */
 /******************************************/
@@ -170,6 +182,76 @@ PyDoc_STRVAR(cards_doc,
 \n\
 List the available card ids.");
 
+
+static PyObject *
+alsacard_list_indexes(PyObject *self, PyObject *args)
+{
+    int rc;
+    int card = -1;
+    PyObject *result = NULL;
+
+    if (!PyArg_ParseTuple(args,":card_indexes"))
+        return NULL;
+
+    result = PyList_New(0);
+
+    for (rc = snd_card_next(&card); !rc && (card >= 0);
+         rc = snd_card_next(&card))
+    {
+        PyObject *item = PyLong_FromLong(card);
+
+        PyList_Append(result, item);
+        Py_DECREF(item);
+    }
+
+    return result;
+}
+
+PyDoc_STRVAR(card_indexes_doc,
+"card_indexes()\n\
+\n\
+List the available card indexes.");
+
+
+static PyObject *
+alsacard_name(PyObject *self, PyObject *args)
+{
+    int err, card;
+    PyObject *result = NULL;
+    char *name = NULL, *longname = NULL;
+
+    if (!PyArg_ParseTuple(args,"i:card_name", &card))
+        return NULL;
+
+    err = snd_card_get_name(card, &name);
+    if (err < 0) {
+        PyErr_Format(ALSAAudioError, "%s [%d]", snd_strerror(err), card);
+        goto exit;
+    }
+
+    err = snd_card_get_longname(card, &longname);
+    if (err < 0) {
+        PyErr_Format(ALSAAudioError, "%s [%d]", snd_strerror(err), card);
+        goto exit;
+    }
+
+    result = PyTuple_New(2);
+    PyTuple_SetItem(result, 0, PyUnicode_FromString(name));
+    PyTuple_SetItem(result, 1, PyUnicode_FromString(longname));
+
+exit:
+    free(name);
+    free(longname);
+
+    return result;
+}
+
+PyDoc_STRVAR(card_name_doc,
+"card_name(card_index) -> Tuple of (name, longname)\n\
+\n\
+Return the card name and long name for card 'card_index'.");
+
+
 static PyObject *
 alsapcm_list(PyObject *self, PyObject *args)
 {
@@ -755,7 +837,11 @@ 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.");
+if no new period has become available since the last call to read.\n\
+\n\
+In case of an overrun, this function will return a negative size: -EPIPE.\n\
+This indicates that data was lost, even if the operation itself succeeded.\n\
+Try using a larger periodsize");
 
 
 static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
@@ -1487,21 +1573,31 @@ Possible values in this list are:\n\
  - 'Capture Exclusive'\n");
 
 
-static int alsamixer_getpercentage(long min, long max, long value)
+static double alsamixer_getpercentage(long min, long max, long value)
 {
     /* Convert from number in range to percentage */
     int range = max - min;
-    int tmp;
 
     if (range == 0)
         return 0;
 
     value -= min;
-    tmp = rint((double)value/(double)range * 100);
-    return tmp;
+    return (double)value/(double)range * 100.0;
 }
 
-static long alsamixer_getphysvolume(long min, long max, int percentage)
+static double alsamixer_getdB(long min, long max, long value)
+{
+    /* Convert from number in range to dB */
+    int range = max - min;
+
+    if (range == 0)
+        return 0;
+
+    value -= min;
+    return log10((double)value/range) * 60.0;
+}
+
+static long alsamixer_getphysvolume(long min, long max, double percentage)
 {
     /* Convert from percentage to number in range */
     int range = max - min;
@@ -1510,57 +1606,76 @@ static long alsamixer_getphysvolume(long min, long max, int percentage)
     if (range == 0)
         return 0;
 
-    tmp = rint((double)range * ((double)percentage*.01)) + min;
+    tmp = rint((double)range * (percentage * .01)) + min;
     return tmp;
 }
 
 static PyObject *
-alsamixer_getvolume(alsamixer_t *self, PyObject *args)
+alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kw)
 {
     snd_mixer_elem_t *elem;
     int channel;
     long ival;
-    PyObject *pcmtypeobj = NULL;
-    long pcmtype;
+    PyObject *dirobj = NULL;
+    long dir;
+    int unit = unit_percent;
     PyObject *result;
     PyObject *item;
 
-    if (!PyArg_ParseTuple(args,"|O:getvolume", &pcmtypeobj))
+    static char *kwlist[] = { "direction", "unit", NULL };
+    
+    if (!PyArg_ParseTupleAndKeywords(args, kw, "|Oi:getvolume", kwlist, &dirobj, &unit))
         return NULL;
 
+    if (unit >= unit_last) {
+        PyErr_SetString(PyExc_ValueError, "unit must be 'percent' or 'dB'");
+        return NULL;
+    }
+    
+    dir = get_pcmtype(dirobj);
+    if (dir < 0) {
+        return NULL;
+    }
+
     if (!self->handle)
     {
         PyErr_SetString(ALSAAudioError, "Mixer is closed");
         return NULL;
     }
 
-    pcmtype = get_pcmtype(pcmtypeobj);
-    if (pcmtype < 0) {
-        return NULL;
-    }
-
     elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
 
     result = PyList_New(0);
 
     for (channel = 0; channel <= SND_MIXER_SCHN_LAST; channel++) {
-        if (pcmtype == SND_PCM_STREAM_PLAYBACK &&
+        if (dir == SND_PCM_STREAM_PLAYBACK &&
             snd_mixer_selem_has_playback_channel(elem, channel))
         {
             snd_mixer_selem_get_playback_volume(elem, channel, &ival);
-            item = PyLong_FromLong(alsamixer_getpercentage(self->pmin,
-                                                           self->pmax,
-                                                           ival));
+            if (unit == unit_percent) {
+                item = PyFloat_FromDouble(
+                    alsamixer_getpercentage(self->pmin, self->pmax, ival));
+            }
+            else {
+                item = PyFloat_FromDouble(
+                    alsamixer_getdB(self->pmin, self->pmax, ival));
+            }
             PyList_Append(result, item);
             Py_DECREF(item);
         }
-        else if (pcmtype == SND_PCM_STREAM_CAPTURE
+        else if (dir == SND_PCM_STREAM_CAPTURE
                  && snd_mixer_selem_has_capture_channel(elem, channel)
-                 && snd_mixer_selem_has_capture_volume(elem)) {
+                 && snd_mixer_selem_has_capture_volume(elem))
+        {
             snd_mixer_selem_get_capture_volume(elem, channel, &ival);
-            item = PyLong_FromLong(alsamixer_getpercentage(self->cmin,
-                                                           self->cmax,
-                                                           ival));
+            if (unit == unit_percent) {
+                item = PyFloat_FromDouble(
+                    alsamixer_getpercentage(self->cmin, self->cmax, ival));
+            }
+            else {
+                item = PyFloat_FromDouble(
+                    alsamixer_getdB(self->cmin, self->cmax, ival));
+            }
             PyList_Append(result, item);
             Py_DECREF(item);
         }
@@ -1570,15 +1685,17 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args)
 }
 
 PyDoc_STRVAR(getvolume_doc,
-"getvolume([pcmtype]) -> List of volume settings (int)\n\
+"getvolume(direction=PCM_PLAYBACK, unit=Percent) -> List of volume settings (float)\n\
 \n\
 Returns a list with the current volume settings for each channel.\n\
-The list elements are integer percentages.\n\
+The list elements are float percentages.\n\
 \n\
-The optional 'pcmtype' argument can be either PCM_PLAYBACK or\n\
+The 'direction' argument can be either PCM_PLAYBACK or\n\
 PCM_CAPTURE, which is relevant if the mixer can control both\n\
 playback and capture volume. The default value is PCM_PLAYBACK\n\
-if the mixer has this capability, otherwise PCM_CAPTURE");
+if the mixer has this capability, otherwise PCM_CAPTURE\
+\n\
+The optional 'unit' argument can be either 'percent' or 'dB'.");
 
 
 static PyObject *
@@ -1744,6 +1861,60 @@ PyDoc_STRVAR(getenum_doc,
 Returns a a tuple. The first element is name of the active enumerated item, \n\
 the second a list available enumerated items.");
 
+
+static PyObject *
+alsamixer_setenum(alsamixer_t *self, PyObject *args)
+{
+    snd_mixer_elem_t *elem;
+    int index, count, rc;
+
+    if (!PyArg_ParseTuple(args, "i:setenum", &index))
+        return NULL;
+
+    if (!self->handle)
+    {
+        PyErr_SetString(ALSAAudioError, "Mixer is closed");
+        return NULL;
+    }
+
+    elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
+    if (!snd_mixer_selem_is_enumerated(elem)) {
+        PyErr_SetString(ALSAAudioError, "Not an enumerated control");
+        return NULL;
+    }
+
+    count = snd_mixer_selem_get_enum_items(elem);
+    if (count < 0)
+    {
+        PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(count),
+                     self->cardname);
+        return NULL;
+    }
+
+    if (index < 0 || index >= count) {
+        PyErr_Format(ALSAAudioError, "Enum index out of range 0 <= %d < %d",
+                     index, count);
+        return NULL;
+    }
+
+    rc = snd_mixer_selem_set_enum_item(elem, 0, index);
+    if (rc)
+    {
+        PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(rc),
+                     self->cardname);
+        return NULL;
+    }
+
+    Py_RETURN_NONE;
+}
+
+PyDoc_STRVAR(setenum_doc,
+"setenum(index) -> None\n\
+\n\
+Sets the value of the enum, where 'index' is an index into the list of\n\
+available enumerated items returned by getenum().");
+
+
 static PyObject *
 alsamixer_getmute(alsamixer_t *self, PyObject *args)
 {
@@ -1850,29 +2021,53 @@ This method will fail if the mixer has no capture switch capabilities.");
 
 
 static PyObject *
-alsamixer_setvolume(alsamixer_t *self, PyObject *args)
+alsamixer_setvolume(alsamixer_t *self, PyObject *args, PyObject *kw)
 {
     snd_mixer_elem_t *elem;
     int i;
-    long volume;
+    double volume = 0.0;
+    PyObject *volumeobj = NULL;
     int physvolume;
-    PyObject *pcmtypeobj = NULL;
-    long pcmtype;
+    PyObject *dirobj = NULL;
+    long dir;
+    int unit = unit_percent; 
     int channel = MIXER_CHANNEL_ALL;
     int done = 0;
 
-    if (!PyArg_ParseTuple(args,"l|iO:setvolume", &volume, &channel,
-                          &pcmtypeobj))
-        return NULL;
-
-    if (volume < 0 || volume > 100)
-    {
-        PyErr_SetString(ALSAAudioError, "Volume must be between 0 and 100");
+    static char *kwlist[] = { "channel", "direction", "unit", NULL };
+    
+    
+    if (!PyArg_ParseTupleAndKeywords(args, kw, "O|iOi:setvolume", kwlist, &volumeobj, &channel,
+                                     &dirobj, &unit)) {
         return NULL;
     }
 
-    pcmtype = get_pcmtype(pcmtypeobj);
-    if (pcmtype < 0) {
+    // unit 
+    if (unit >= unit_last) {
+        PyErr_SetString(PyExc_ValueError, "unit must be 'percent' or 'dB'");
+        return NULL;
+    }
+
+    if (PyLong_Check(volumeobj)) {
+        volume = (double)PyLong_AsLong(volumeobj);
+    }
+    else if (PyFloat_Check(volumeobj)) {
+        volume = PyFloat_AsDouble(volumeobj);
+    }
+    else {
+        PyErr_SetString(PyExc_ValueError, "Volume must be integer or float");
+        return NULL;
+    }
+    
+    if (unit == unit_percent && (volume < 0.0 || volume > 100.0))
+    {
+        PyErr_SetString(PyExc_ValueError, "Volume in percent must be between 0 and 100");
+        return NULL;
+    }
+
+    // pcmtype
+    dir = get_pcmtype(dirobj);
+    if (dir < 0) {
         return NULL;
     }
 
@@ -1881,41 +2076,51 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args)
         PyErr_SetString(ALSAAudioError, "Mixer is closed");
         return NULL;
     }
-
+    
     elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
 
-    if (!pcmtypeobj || (pcmtypeobj == Py_None))
+    if (!dirobj || (dirobj == Py_None))
     {
         if (self->pchannels)
-            pcmtype = SND_PCM_STREAM_PLAYBACK;
+            dir = SND_PCM_STREAM_PLAYBACK;
         else
-            pcmtype = SND_PCM_STREAM_CAPTURE;
+            dir = SND_PCM_STREAM_CAPTURE;
     }
 
     for (i = 0; i <= SND_MIXER_SCHN_LAST; i++)
     {
         if (channel == -1 || channel == i)
         {
-            if (pcmtype == SND_PCM_STREAM_PLAYBACK &&
+            if (dir == SND_PCM_STREAM_PLAYBACK &&
                 snd_mixer_selem_has_playback_channel(elem, i)) {
-                physvolume = alsamixer_getphysvolume(self->pmin,
-                                                     self->pmax, volume);
-                snd_mixer_selem_set_playback_volume(elem, i, physvolume);
+                if (unit == unit_percent) {
+                    physvolume = alsamixer_getphysvolume(self->pmin,
+                                                         self->pmax, volume);
+                    snd_mixer_selem_set_playback_volume(elem, i, physvolume);
+                }
+                else {
+                    snd_mixer_selem_set_playback_dB(elem, i, (long)(volume * 100.0), 0);
+                }
                 done++;
             }
-            else if (pcmtype == SND_PCM_STREAM_CAPTURE
+            else if (dir == SND_PCM_STREAM_CAPTURE
                      && snd_mixer_selem_has_capture_channel(elem, i)
                      && snd_mixer_selem_has_capture_volume(elem))
             {
-                physvolume = alsamixer_getphysvolume(self->cmin, self->cmax,
-                                                     volume);
-                snd_mixer_selem_set_capture_volume(elem, i, physvolume);
+                if (unit == unit_percent) {
+                    physvolume = alsamixer_getphysvolume(self->cmin, self->cmax,
+                                                         volume);
+                    snd_mixer_selem_set_capture_volume(elem, i, physvolume);
+                }
+                else {
+                    snd_mixer_selem_set_capture_dB(elem, i, (long)(volume * 100), 0);
+                }
                 done++;
             }
         }
     }
 
-    if(!done)
+    if (!done)
     {
         PyErr_Format(ALSAAudioError, "No such channel [%s]",
                      self->cardname);
@@ -1927,19 +2132,21 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args)
 }
 
 PyDoc_STRVAR(setvolume_doc,
-"setvolume(volume[[, channel] [, pcmtype]])\n\
+"setvolume(volume, channel=MIXER_CHANNEL_ALL, direction=PCM_PLAYBACK, unit='percent')\n\
 \n\
 Change the current volume settings for this mixer. The volume argument\n\
-controls the new volume setting as an integer percentage.\n\
+controls the new volume setting as a percentage.\n\
 If the optional argument channel is present, the volume is set only for\n\
 this channel. This assumes that the mixer can control the volume for the\n\
 channels independently.\n\
 \n\
-The optional direction argument can be either PCM_PLAYBACK or PCM_CAPTURE.\n\
+The optional 'direction' argument can be either PCM_PLAYBACK or PCM_CAPTURE.\n\
 It is relevant if the mixer has independent playback and capture volume\n\
 capabilities, and controls which of the volumes will be changed.\n\
-The default is 'playback' if the mixer has this capability, otherwise\n\
-'capture'.");
+The default is PCM_PLAYBACK if the mixer has this capability, otherwise\n\
+PCM_CAPTURE.\n\
+\n\
+The optional 'unit' argument can be either 'percent' (the default) or 'dB'.");
 
 
 static PyObject *
@@ -1950,6 +2157,7 @@ alsamixer_setmute(alsamixer_t *self, PyObject *args)
     int mute = 0;
     int done = 0;
     int channel = MIXER_CHANNEL_ALL;
+    
     if (!PyArg_ParseTuple(args,"i|i:setmute", &mute, &channel))
         return NULL;
 
@@ -2115,6 +2323,38 @@ PyDoc_STRVAR(polldescriptors_doc,
 Return a list of file descriptors and event masks\n\
 suitable for use with poll to monitor changes on this mixer.");
 
+static PyObject *
+alsamixer_handleevents(alsamixer_t *self, PyObject *args)
+{
+    int handled;
+
+    if (!PyArg_ParseTuple(args,":handleevents"))
+        return NULL;
+
+    if (!self->handle)
+    {
+        PyErr_SetString(ALSAAudioError, "Mixer is closed");
+        return NULL;
+    }
+
+    handled = snd_mixer_handle_events(self->handle);
+    if (handled < 0)
+    {
+        PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(handled),
+                     self->cardname);
+        return NULL;
+    }
+
+    return PyLong_FromLong(handled);
+}
+
+PyDoc_STRVAR(handleevents_doc,
+"handleevents() -> int\n\
+\n\
+Acknowledge events on the polldescriptors() file descriptors\n\
+to prevent subsequent polls from returning the same events again.\n\
+Returns the number of events that were acknowledged.");
+
 static PyMethodDef alsamixer_methods[] = {
     {"cardname", (PyCFunction)alsamixer_cardname, METH_VARARGS,
      mixer_cardname_doc},
@@ -2125,18 +2365,21 @@ static PyMethodDef alsamixer_methods[] = {
      switchcap_doc},
     {"volumecap", (PyCFunction)alsamixer_volumecap, METH_VARARGS,
      volumecap_doc},
-    {"getvolume", (PyCFunction)alsamixer_getvolume, METH_VARARGS,
+    {"getvolume", (PyCFunction)alsamixer_getvolume, METH_VARARGS | METH_KEYWORDS,
      getvolume_doc},
     {"getrange", (PyCFunction)alsamixer_getrange, METH_VARARGS, getrange_doc},
     {"getenum", (PyCFunction)alsamixer_getenum, METH_VARARGS, getenum_doc},
     {"getmute", (PyCFunction)alsamixer_getmute, METH_VARARGS, getmute_doc},
     {"getrec", (PyCFunction)alsamixer_getrec, METH_VARARGS, getrec_doc},
-    {"setvolume", (PyCFunction)alsamixer_setvolume, METH_VARARGS,
+    {"setvolume", (PyCFunction)alsamixer_setvolume, METH_VARARGS | METH_KEYWORDS,
      setvolume_doc},
+    {"setenum", (PyCFunction)alsamixer_setenum, METH_VARARGS, setenum_doc},
     {"setmute", (PyCFunction)alsamixer_setmute, METH_VARARGS, setmute_doc},
     {"setrec", (PyCFunction)alsamixer_setrec, METH_VARARGS, setrec_doc},
     {"polldescriptors", (PyCFunction)alsamixer_polldescriptors, METH_VARARGS,
      polldescriptors_doc},
+    {"handleevents", (PyCFunction)alsamixer_handleevents, METH_VARARGS,
+     handleevents_doc},
 
     {NULL, NULL}
 };
@@ -2201,6 +2444,8 @@ static PyTypeObject ALSAMixerType = {
 /******************************************/
 
 static PyMethodDef alsaaudio_methods[] = {
+    { "card_indexes", (PyCFunction)alsacard_list_indexes, METH_VARARGS, card_indexes_doc},
+    { "card_name", (PyCFunction)alsacard_name, METH_VARARGS, card_name_doc},
     { "cards", (PyCFunction)alsacard_list, METH_VARARGS, cards_doc},
     { "pcms", (PyCFunction)alsapcm_list, METH_VARARGS, pcms_doc},
     { "mixers", (PyCFunction)alsamixer_list, METH_VARARGS|METH_KEYWORDS, mixers_doc},
@@ -2276,6 +2521,9 @@ PyObject *PyInit_alsaaudio(void)
     Py_INCREF(ALSAAudioError);
     PyModule_AddObject(m, "ALSAAudioError", ALSAAudioError);
 
+    _EXPORT_INT(m, "Percent", unit_percent);
+    _EXPORT_INT(m, "dB", unit_dB);
+
     _EXPORT_INT(m, "PCM_PLAYBACK",SND_PCM_STREAM_PLAYBACK);
     _EXPORT_INT(m, "PCM_CAPTURE",SND_PCM_STREAM_CAPTURE);
 
@@ -2307,6 +2555,22 @@ PyObject *PyInit_alsaaudio(void)
     _EXPORT_INT(m, "PCM_FORMAT_IMA_ADPCM",SND_PCM_FORMAT_IMA_ADPCM);
     _EXPORT_INT(m, "PCM_FORMAT_MPEG",SND_PCM_FORMAT_MPEG);
     _EXPORT_INT(m, "PCM_FORMAT_GSM",SND_PCM_FORMAT_GSM);
+    _EXPORT_INT(m, "PCM_FORMAT_S24_3LE",SND_PCM_FORMAT_S24_3LE);
+    _EXPORT_INT(m, "PCM_FORMAT_S24_3BE",SND_PCM_FORMAT_S24_3BE);
+    _EXPORT_INT(m, "PCM_FORMAT_U24_3LE",SND_PCM_FORMAT_U24_3LE);
+    _EXPORT_INT(m, "PCM_FORMAT_U24_3BE",SND_PCM_FORMAT_U24_3BE);
+
+     /* DSD sample formats are included in ALSA 1.0.29 and higher
+      * define OVERRIDE_DSD_COMPILE to include DSD sample support
+      * if you use a patched ALSA lib version
+      */
+
+#if SND_LIB_VERSION >= 0x1001d || defined OVERRIDE_DSD_COMPILE
+    _EXPORT_INT(m, "PCM_FORMAT_DSD_U8", SND_PCM_FORMAT_DSD_U8);
+    _EXPORT_INT(m, "PCM_FORMAT_DSD_U16_LE", SND_PCM_FORMAT_DSD_U16_LE);
+    _EXPORT_INT(m, "PCM_FORMAT_DSD_U32_LE", SND_PCM_FORMAT_DSD_U32_LE);
+    _EXPORT_INT(m, "PCM_FORMAT_DSD_U32_BE", SND_PCM_FORMAT_DSD_U32_BE);
+#endif
 
     /* Mixer stuff */
     _EXPORT_INT(m, "MIXER_CHANNEL_ALL", MIXER_CHANNEL_ALL);

doc/conf.py

@@ -1,182 +1,160 @@
 # -*- coding: utf-8 -*-
 #
-# alsaaudio documentation build configuration file, created by
-# sphinx-quickstart on Sat Nov 22 00:17:09 2008.
+# alsaaudio documentation documentation build configuration file, created by
+# sphinx-quickstart on Thu Mar 30 23:52:21 2017.
 #
-# This file is execfile()d with the current directory set to its containing dir.
+# This file is execfile()d with the current directory set to its
+# containing dir.
 #
-# The contents of this file are pickled, so don't put values in the namespace
-# that aren't pickleable (module imports are okay, they're removed automatically).
+# Note that not all possible configuration values are present in this
+# autogenerated file.
 #
-# All configuration values have a default value; values that are commented out
-# serve to show the default value.
+# All configuration values have a default; values that are commented out
+# serve to show the default.
 
-import sys, os
+# If extensions (or modules to document with autodoc) are in another directory,
+# add these directories to sys.path here. If the directory is relative to the
+# documentation root, use os.path.abspath to make it absolute, like shown here.
+#
+# import os
+# import sys
+# sys.path.insert(0, os.path.abspath('.'))
 
-sys.path.append('..')
+import sys
+sys.path.insert(0, '..')
 from setup import pyalsa_version
 
-# If your extensions are in another directory, add it here. If the directory
-# is relative to the documentation root, use os.path.abspath to make it
-# absolute, like shown here.
-#sys.path.append(os.path.abspath('some/directory'))
 
-# General configuration
-# ---------------------
+# -- General configuration ------------------------------------------------
 
-# Add any Sphinx extension module names here, as strings. They can be extensions
-# coming with Sphinx (named 'sphinx.ext.*') or your custom ones.
+# If your documentation needs a minimal Sphinx version, state it here.
+#
+# needs_sphinx = '1.0'
+
+# Add any Sphinx extension module names here, as strings. They can be
+# extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
+# ones.
 extensions = []
 
 # Add any paths that contain templates here, relative to this directory.
-templates_path = ['.templates']
+templates_path = ['_templates']
 
-# The suffix of source filenames.
+# The suffix(es) of source filenames.
+# You can specify multiple suffix as a list of string:
+#
+# source_suffix = ['.rst', '.md']
 source_suffix = '.rst'
 
 # The master toctree document.
 master_doc = 'index'
 
-# General substitutions.
-project = u'alsaaudio'
-copyright = u'2008-2009, Casper Wilstrup, Lars Immisch'
+# General information about the project.
+project = u'alsaaudio documentation'
+copyright = u'2017, Lars Immisch & Casper Wilstrup'
+author = u'Lars Immisch & Casper Wilstrup'
 
-# The default replacements for |version| and |release|, also used in various
-# other places throughout the built documents.
+# The version info for the project you're documenting, acts as replacement for
+# |version| and |release|, also used in various other places throughout the
+# built documents.
 #
 # The short X.Y version.
 version = pyalsa_version
 # The full version, including alpha/beta/rc tags.
-release = pyalsa_version
+release = version
 
-# There are two options for replacing |today|: either, you set today to some
-# non-false value, then it is used:
-#today = ''
-# Else, today_fmt is used as the format for a strftime call.
-today_fmt = '%B %d, %Y'
+# The language for content autogenerated by Sphinx. Refer to documentation
+# for a list of supported languages.
+#
+# This is also used if you do content translation via gettext catalogs.
+# Usually you set "language" from the command line for these cases.
+language = None
 
-# List of documents that shouldn't be included in the build.
-#unused_docs = []
-
-# List of directories, relative to source directories, that shouldn't be searched
-# for source files.
-exclude_trees = ['.build']
-
-# The reST default role (used for this markup: `text`) to use for all documents.
-#default_role = None
-
-# If true, '()' will be appended to :func: etc. cross-reference text.
-#add_function_parentheses = True
-
-# If true, the current module name will be prepended to all description
-# unit titles (such as .. function::).
-#add_module_names = True
-
-# If true, sectionauthor and moduleauthor directives will be shown in the
-# output. They are ignored by default.
-#show_authors = False
+# List of patterns, relative to source directory, that match files and
+# directories to ignore when looking for source files.
+# This patterns also effect to html_static_path and html_extra_path
+exclude_patterns = ['_build', 'Thumbs.db', '.DS_Store']
 
 # The name of the Pygments (syntax highlighting) style to use.
 pygments_style = 'sphinx'
 
+# If true, `todo` and `todoList` produce output, else they produce nothing.
+todo_include_todos = False
 
-# Options for HTML output
-# -----------------------
 
-# The style sheet to use for HTML and HTML Help pages. A file of that name
-# must exist either in Sphinx' static/ path, or in one of the custom paths
-# given in html_static_path.
-html_style = 'default.css'
+# -- Options for HTML output ----------------------------------------------
 
-# The name for this set of Sphinx documents.  If None, it defaults to
-# "<project> v<release> documentation".
-#html_title = None
+# The theme to use for HTML and HTML Help pages.  See the documentation for
+# a list of builtin themes.
+#
+html_theme = 'alabaster'
 
-# A shorter title for the navigation bar.  Default is the same as html_title.
-#html_short_title = None
-
-# The name of an image file (relative to this directory) to place at the top
-# of the sidebar.
-#html_logo = None
-
-# The name of an image file (within the static path) to use as favicon of the
-# docs.  This file should be a Windows icon file (.ico) being 16x16 or 32x32
-# pixels large.
-#html_favicon = None
+# Theme options are theme-specific and customize the look and feel of a theme
+# further.  For a list of options available for each theme, see the
+# documentation.
+#
+# html_theme_options = {}
 
 # Add any paths that contain custom static files (such as style sheets) here,
 # relative to this directory. They are copied after the builtin static files,
 # so a file named "default.css" will overwrite the builtin "default.css".
-html_static_path = ['static']
+html_static_path = ['_static']
 
-# If not '', a 'Last updated on:' timestamp is inserted at every page bottom,
-# using the given strftime format.
-html_last_updated_fmt = '%b %d, %Y'
 
-# If true, SmartyPants will be used to convert quotes and dashes to
-# typographically correct entities.
-#html_use_smartypants = True
-
-# Custom sidebar templates, maps document names to template names.
-#html_sidebars = {}
-
-# Additional templates that should be rendered to pages, maps page names to
-# template names.
-#html_additional_pages = {}
-
-# If false, no module index is generated.
-#html_use_modindex = True
-
-# If false, no index is generated.
-#html_use_index = True
-
-# If true, the index is split into individual pages for each letter.
-#html_split_index = False
-
-# If true, the reST sources are included in the HTML build as _sources/<name>.
-#html_copy_source = True
-
-# If true, an OpenSearch description file will be output, and all pages will
-# contain a <link> tag referring to it.  The value of this option must be the
-# base URL from which the finished HTML is served.
-#html_use_opensearch = ''
-
-# If nonempty, this is the file name suffix for HTML files (e.g. ".xhtml").
-#html_file_suffix = ''
+# -- Options for HTMLHelp output ------------------------------------------
 
 # Output file base name for HTML help builder.
-htmlhelp_basename = 'alsaaudiodoc'
+htmlhelp_basename = 'alsaaudiodocumentationdoc'
 
 
-# Options for LaTeX output
-# ------------------------
+# -- Options for LaTeX output ---------------------------------------------
 
-# The paper size ('letter' or 'a4').
-#latex_paper_size = 'letter'
+latex_elements = {
+    # The paper size ('letterpaper' or 'a4paper').
+    #
+    # 'papersize': 'letterpaper',
 
-# The font size ('10pt', '11pt' or '12pt').
-#latex_font_size = '10pt'
+    # The font size ('10pt', '11pt' or '12pt').
+    #
+    # 'pointsize': '10pt',
+
+    # Additional stuff for the LaTeX preamble.
+    #
+    # 'preamble': '',
+
+    # Latex figure (float) alignment
+    #
+    # 'figure_align': 'htbp',
+}
 
 # Grouping the document tree into LaTeX files. List of tuples
-# (source start file, target name, title, author, document class [howto/manual]).
+# (source start file, target name, title,
+#  author, documentclass [howto, manual, or own class]).
 latex_documents = [
-  ('index', 'alsaaudio.tex', u'alsaaudio Documentation',
-   u'Casper Wilstrup, Lars Immisch', 'manual'),
+    (master_doc, 'alsaaudiodocumentation.tex', u'alsaaudio documentation Documentation',
+     u'Lars Immisch', 'manual'),
 ]
 
-# The name of an image file (relative to this directory) to place at the top of
-# the title page.
-#latex_logo = None
 
-# For "manual" documents, if this is true, then toplevel headings are parts,
-# not chapters.
-#latex_use_parts = False
+# -- Options for manual page output ---------------------------------------
+
+# One entry per manual page. List of tuples
+# (source start file, name, description, authors, manual section).
+man_pages = [
+    (master_doc, 'alsaaudiodocumentation', u'alsaaudio documentation Documentation',
+     [author], 1)
+]
+
+
+# -- Options for Texinfo output -------------------------------------------
+
+# Grouping the document tree into Texinfo files. List of tuples
+# (source start file, target name, title, author,
+#  dir menu entry, description, category)
+texinfo_documents = [
+    (master_doc, 'alsaaudiodocumentation', u'alsaaudio documentation Documentation',
+     author, 'alsaaudiodocumentation', 'One line description of project.',
+     'Miscellaneous'),
+]
 
-# Additional stuff for the LaTeX preamble.
-#latex_preamble = ''
 
-# Documents to append as an appendix to all manuals.
-#latex_appendices = []
 
-# If false, no module index is generated.
-#latex_use_modindex = True

doc/index.rst

@@ -1,22 +1,32 @@
+.. alsaaudio documentation documentation master file, created by
+   sphinx-quickstart on Thu Mar 30 23:52:21 2017.
+   You can adapt this file completely to your liking, but it should at least
+   contain the root `toctree` directive.
+
 alsaaudio documentation
-=======================
+===================================================
 
 .. toctree::
    :maxdepth: 2
+   :caption: Contents:
 
    pyalsaaudio
    terminology
    libalsaaudio
 
+Download
+========
 
-Github pages
-=================
-
-* `Project page <https://github.com/larsimmisch/pyalsaaudio/>`_
 * `Download from pypi <https://pypi.python.org/pypi/pyalsaaudio>`_
+  
+      
+Github
+======
+
+* `Repository <https://github.com/larsimmisch/pyalsaaudio/>`_
 * `Bug tracker <https://github.com/larsimmisch/pyalsaaudio/issues>`_
 
-
+ 
 Indices and tables
 ==================
 
@@ -24,3 +34,5 @@ Indices and tables
 * :ref:`modindex`
 * :ref:`search`
 
+    
+  

doc/libalsaaudio.rst

@@ -193,10 +193,10 @@ PCM objects have the following methods:
    ``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_S24_LE``      Signed 24 bit samples for each channel (Little Endian byte order in 4 bytes)
+   ``PCM_FORMAT_S24_BE``      Signed 24 bit samples for each channel (Big Endian byte order in 4 bytes)
+   ``PCM_FORMAT_U24_LE``      Unsigned 24 bit samples for each channel (Little Endian byte order in 4 bytes)
+   ``PCM_FORMAT_U24_BE``      Unsigned 24 bit samples for each channel (Big Endian byte order in 4 bytes)
    ``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)
@@ -210,6 +210,10 @@ PCM objects have the following methods:
    ``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
+   ``PCM_FORMAT_S24_3LE``     Signed 24 bit samples for each channel (Little Endian byte order in 3 bytes)
+   ``PCM_FORMAT_S24_3BE``     Signed 24 bit samples for each channel (Big Endian byte order in 3 bytes)
+   ``PCM_FORMAT_U24_3LE``     Unsigned 24 bit samples for each channel (Little Endian byte order in 3 bytes)
+   ``PCM_FORMAT_U24_3BE``     Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
    =========================  ===============
    
 
@@ -233,6 +237,9 @@ PCM objects have the following methods:
    ``(0,'')`` if no new period has become available since the last
    call to read.
 
+   In case of an overrun, this function will return a negative size: :const:`-EPIPE`.
+   This indicates that data was lost, even if the operation itself succeeded.
+   Try using a larger periodsize.
 
 .. method:: PCM.write(data)
 
@@ -256,6 +263,17 @@ PCM objects have the following methods:
    If *enable* is :const:`True`, playback or capture is paused.
    Otherwise, playback/capture is resumed.
 
+
+.. method:: PCM.polldescriptors()
+
+   Returns a tuple of *(file descriptor, eventmask)* that can be used to
+   wait for changes on the mixer with *select.poll*.
+
+   The *eventmask* value is compatible with `poll.register`__ in the Python 
+   :const:`select` module.
+
+__ poll_objects_
+
 **A few hints on using PCM devices for playback**
 
 The most common reason for problems with playback of PCM audio is that writes 
@@ -425,31 +443,35 @@ Mixer objects have the following methods:
    This method will fail if the mixer has no capture switch capabilities.
 
 
-.. method:: Mixer.getvolume([direction])
+.. method:: Mixer.getvolume(direction=PCM_PLAYBACK, unit=Percent)
 
    Returns a list with the current volume settings for each channel. The list
-   elements are integer percentages.
+   elements are percentages or dB values, depending on *unit*.
 
-   The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
+   The *direction* argument can be either :const:`PCM_PLAYBACK` or
    :const:`PCM_CAPTURE`, which is relevant if the mixer can control both
    playback and capture volume. The default value is :const:`PCM_PLAYBACK`
    if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
 
 
-.. method:: Mixer.setvolume(volume, [channel], [direction])
+.. method:: Mixer.setvolume(volume, channel=MIXER_CHANNEL_ALL, direction=PCM_PLAYBACK, unit=Percent)
 
    Change the current volume settings for this mixer. The *volume* argument
-   controls the new volume setting as an integer percentage.
+   controls the new volume setting as either a percentage or a dB value. Both
+   integer and floating point values can be given.
 
-   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 *channel* argument can be used to restrict the channels for which the volume is
+   set. By default, the volume of all channels is adjusted. This assumes that the mixer
+   can control the volume for the channels independently.
 
-   The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
+   The *direction* argument can be either :const:`PCM_PLAYBACK` or
    :const:`PCM_CAPTURE`, which is relevant if the mixer can control both
    playback and capture volume. The default value is :const:`PCM_PLAYBACK`
    if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
 
+   The *unit* argument determines how the volume value is interpreted, as a prcentage
+   or as a dB value.
+   
 .. method:: Mixer.setmute(mute, [channel])
 
    Sets the mute flag to a new value. The *mute* argument is either 0 for not
@@ -473,9 +495,20 @@ Mixer objects have the following methods:
 
 .. method:: Mixer.polldescriptors()
 
-   Returns a tuple of (file descriptor, eventmask) that can be used to
+   Returns a tuple of *(file descriptor, eventmask)* that can be used to
    wait for changes on the mixer with *select.poll*.
 
+   The *eventmask* value is compatible with `poll.register`__ in the Python 
+   :const:`select` module.
+
+__ poll_objects_
+
+.. method:: Mixer.handleevents()
+
+   Acknowledge events on the *polldescriptors* file descriptors
+   to prevent subsequent polls from returning the same events again.
+   Returns the number of events that were acknowledged.
+
 **A rant on the ALSA Mixer API**
 
 The ALSA mixer API is extremely complicated - and hardly documented at all.
@@ -614,3 +647,5 @@ argument::
 .. rubric:: Footnotes
 
 .. [#f1]   ALSA also allows ``PCM_ASYNC``, but this is not supported yet.
+
+.. _poll_objects: http://docs.python.org/library/select.html#poll-objects

doc/pyalsaaudio.rst

@@ -33,7 +33,7 @@ wish (even commercial purposes). There is no warranty whatsoever.
    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. 
+   If you find bugs in the wrappers please use thegithub issue 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.
 
@@ -110,25 +110,32 @@ And then as root:  ---   ::
 Testing
 *******
 
-First of all, run::
-   
-   $ python test.py
+Make sure that :code:`aplay` plays a file through the soundcard you want, then
+try::
 
-This is a small test suite that mostly performs consistency tests. If
-it fails, please file a `bug report
-<https://github.com/larsimmisch/pyalsaaudio/issues>`_.
+  $ python playwav.py <filename.wav>
+
+If :code:`aplay` needs a device argument, like
+:code:`aplay -D hw:CARD=sndrpihifiberry,DEV=0`, use::
+
+   $ python playwav.py -d hw:CARD=sndrpihifiberry,DEV=0 <filename.wav>
 
 To test PCM recordings (on your default soundcard), verify your
 microphone works, then do::
 
-   $ python recordtest.py <filename>
+   $ python recordtest.py -d <device> <filename>
 
 Speak into the microphone, and interrupt the recording at any time
 with ``Ctl-C``.
 
 Play back the recording with::
 
-   $ python playbacktest.py <filename>
+   $ python playbacktest.py-d <device>  <filename>
 
+There is a minimal test suite in :code:`test.py`, but it is
+a bit dependent on the ALSA configuration and may fail without indicating
+a real problem.
 
+If you find bugs/problems, please file a `bug report
+<https://github.com/larsimmisch/pyalsaaudio/issues>`_.
 

doc/terminology.rst

@@ -46,7 +46,7 @@ Data rate
 
    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)
+   kb/s (almost 5 MB sound data per second)
 
 Period
    When the hardware processes data this is done in chunks of frames. The time

isine.py

@@ -6,30 +6,50 @@
 
 from __future__ import print_function
 
+import sys
 from threading import Thread
-from queue import Queue, Empty
+from multiprocessing import Queue
+
+if sys.version_info[0] < 3:
+    from Queue import Empty
+else:
+    from queue import Empty
+
 from math import pi, sin
 import struct
 import alsaaudio
 
-sampling_rate = 44100
+sampling_rate = 48000
+
 format = alsaaudio.PCM_FORMAT_S16_LE
 framesize = 2 # bytes per frame for the values above
+channels = 2
 
-def digitize(s):
-    if s > 1.0 or s < -1.0:
-        raise ValueError
-        
-    return struct.pack('h', int(s * 32767)) 
+def nearest_frequency(frequency):
+    # calculate the nearest frequency where the wave form fits into the buffer
+    # in other words, select f so that sampling_rate/f is an integer
+    return float(sampling_rate)/int(sampling_rate/frequency)
 
-def generate(frequency):
-    # generate a buffer with a sine wave of frequency
-    size = int(sampling_rate / frequency)
-    buffer = bytes()
-    for i in range(size):
-        buffer = buffer + digitize(sin(i/(2 * pi)))
+def generate(frequency, duration = 0.125):
+    # generate a buffer with a sine wave of `frequency`
+    # that is approximately `duration` seconds long
 
-    return buffer
+    # the buffersize we approximately want
+    target_size = int(sampling_rate * channels * duration)
+
+    # the length of a full sine wave at the frequency
+    cycle_size = int(sampling_rate / frequency)
+
+    # number of full cycles we can fit into target_size
+    factor = int(target_size / cycle_size)
+
+    size = max(int(cycle_size * factor), 1)
+    
+    sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \
+             for i in range(size)]
+             
+    return struct.pack('%dh' % size, *sine)
+                                                                                  
 
 class SinePlayer(Thread):
     
@@ -37,7 +57,7 @@ class SinePlayer(Thread):
         Thread.__init__(self)
         self.setDaemon(True)
         self.device = alsaaudio.PCM()
-        self.device.setchannels(1)
+        self.device.setchannels(channels)
         self.device.setformat(format)
         self.device.setrate(sampling_rate)
         self.queue = Queue()
@@ -47,19 +67,15 @@ class SinePlayer(Thread):
         '''This is called outside of the player thread'''
         # we generate the buffer in the calling thread for less
         # latency when switching frequencies
-        
 
-        # More than 100 writes/s are pushing it - play multiple buffers
-        # for higher frequencies
+        if frequency > sampling_rate / 2:
+            raise ValueError('maximum frequency is %d' % (sampling_rate / 2))
 
-        factor = int(frequency/100.0)
-        if factor == 0:
-            factor = 1
-        
-        buf = generate(frequency) * factor
-        print('factor: %d, frames: %d' % (factor, len(buf) / framesize))
+        f = nearest_frequency(frequency)
+        print('nearest frequency: %f' % f)
 
-        self.queue.put( buf)
+        buf = generate(f)
+        self.queue.put(buf)
                         
     def run(self):
         buffer = None

mixertest.py

@@ -23,6 +23,12 @@ import sys
 import getopt
 import alsaaudio
 
+def list_cards():
+    print("Available sound cards:")
+    for i in alsaaudio.card_indexes():
+        (name, longname) = alsaaudio.card_name(i)
+        print("  %d: %s (%s)" % (i, name, longname))
+
 def list_mixers(kwargs):
     print("Available mixer controls:")
     for m in alsaaudio.mixers(**kwargs):
@@ -40,13 +46,17 @@ def show_mixer(name, kwargs):
     print("Capabilities: %s %s" % (' '.join(mixer.volumecap()),
                                    ' '.join(mixer.switchcap())))
     volumes = mixer.getvolume()
-    for i in range(len(volumes)):
-        print("Channel %i volume: %i%%" % (i,volumes[i]))
-        
+    for i, v in enumerate(volumes):
+        print("Channel %i volume: %.02f%%" % (i, v))
+
+    volumes = mixer.getvolume(unit=alsaaudio.dB)
+    for i, v in enumerate(volumes):
+        print("Channel %i volume: %.02fdB" % (i, v))
+
     try:
         mutes = mixer.getmute()
-        for i in range(len(mutes)):
-            if mutes[i]:
+        for i, m in enumerate(mutes):
+            if m:
                 print("Channel %i is muted" % i)
     except alsaaudio.ALSAAudioError:
         # May not support muting
@@ -54,8 +64,8 @@ def show_mixer(name, kwargs):
 
     try:
         recs = mixer.getrec()
-        for i in range(len(recs)):
-            if recs[i]:
+        for i, r in enumerate(recs):
+            if r:
                 print("Channel %i is recording" % i)
     except alsaaudio.ALSAAudioError:
         # May not support recording
@@ -113,6 +123,8 @@ if __name__ == '__main__':
         else:
             usage()
 
+    list_cards()
+
     if not len(args):
         list_mixers(kwargs)
     elif len(args) == 1:

playbacktest.py

@@ -21,17 +21,17 @@ import getopt
 import alsaaudio
 
 def usage():
-    print('usage: playbacktest.py [-c <card>] <file>', file=sys.stderr)
+    print('usage: playbacktest.py [-d <device>] <file>', file=sys.stderr)
     sys.exit(2)
 
 if __name__ == '__main__':
 
-    card = 'default'
+    device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'c:')
+    opts, args = getopt.getopt(sys.argv[1:], 'd:')
     for o, a in opts:
-        if o == '-c':
-            card = a
+        if o == '-d':
+            device = a
 
     if not args:
         usage()
@@ -39,7 +39,7 @@ if __name__ == '__main__':
     f = open(args[0], 'rb')
 
     # Open the device in playback mode. 
-    out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, card=card)
+    out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, device=device)
 
     # Set attributes: Mono, 44100 Hz, 16 bit little endian frames
     out.setchannels(1)

playwav.py

@@ -24,39 +24,41 @@ def play(device, f):
     elif f.getsampwidth() == 2:
         device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
     elif f.getsampwidth() == 3:
-        device.setformat(alsaaudio.PCM_FORMAT_S24_LE)
+        device.setformat(alsaaudio.PCM_FORMAT_S24_3LE)
     elif f.getsampwidth() == 4:
         device.setformat(alsaaudio.PCM_FORMAT_S32_LE)
     else:
         raise ValueError('Unsupported format')
 
-    device.setperiodsize(320)
+    periodsize = f.getframerate() / 8
+
+    device.setperiodsize(periodsize)
     
-    data = f.readframes(320)
+    data = f.readframes(periodsize)
     while data:
         # Read data from stdin
         device.write(data)
-        data = f.readframes(320)
+        data = f.readframes(periodsize)
 
 
 def usage():
-    print('usage: playwav.py [-c <card>] <file>', file=sys.stderr)
+    print('usage: playwav.py [-d <device>] <file>', file=sys.stderr)
     sys.exit(2)
 
 if __name__ == '__main__':
 
-    card = 'default'
+    device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'c:')
+    opts, args = getopt.getopt(sys.argv[1:], 'd:')
     for o, a in opts:
-        if o == '-c':
-            card = a
+        if o == '-d':
+            device = a
 
     if not args:
         usage()
         
     f = wave.open(args[0], 'rb')
-    device = alsaaudio.PCM(card=card)
+    device = alsaaudio.PCM(device=device)
 
     play(device, f)
 

recordtest.py

@@ -4,7 +4,7 @@
 ##
 ## This is an example of a simple sound capture script.
 ##
-## The script opens an ALSA pcm forsound capture. Set
+## The script opens an ALSA pcm device for sound capture, sets
 ## various attributes of the capture, and reads in a loop,
 ## writing the data to standard out.
 ##
@@ -22,17 +22,17 @@ import getopt
 import alsaaudio
 
 def usage():
-    print('usage: recordtest.py [-c <card>] <file>', file=sys.stderr)
+    print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr)
     sys.exit(2)
 
 if __name__ == '__main__':
 
-    card = 'default'
+    device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'c:')
+    opts, args = getopt.getopt(sys.argv[1:], 'd:')
     for o, a in opts:
-        if o == '-c':
-            card = a
+        if o == '-d':
+            device = a
 
     if not args:
         usage()
@@ -42,7 +42,7 @@ if __name__ == '__main__':
     # Open the device in nonblocking capture mode. The last argument could
     # just as well have been zero for blocking mode. Then we could have
     # left out the sleep call in the bottom of the loop
-    inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, card)
+    inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, device=device)
 
     # Set attributes: Mono, 44100 Hz, 16 bit little endian samples
     inp.setchannels(1)

setup.py

@@ -4,25 +4,11 @@
 It is fairly complete for PCM devices and Mixer access.
 '''
 
-import subprocess
-from distutils.core import setup
-from distutils.extension import Extension
+from setuptools import setup
+from setuptools.extension import Extension
 from sys import version
 
-def gitrev():
-    rev = subprocess.check_output(['git', 'describe', '--tags', '--dirty=-dev',
-                                  '--always'])
-    return rev.decode('utf-8').strip()
-
-pyalsa_version = gitrev()
-
-# patch distutils if it's too old to cope with the "classifiers" or
-# "download_url" keywords
-from sys import version
-if version < '2.2.3':
-    from distutils.dist import DistributionMetadata
-    DistributionMetadata.classifiers = None
-    DistributionMetadata.download_url = None
+pyalsa_version = '0.8.4'
 
 if __name__ == '__main__':
     setup(

test.py

@@ -44,16 +44,11 @@ class MixerTest(unittest.TestCase):
     def testMixer(self):
         """Open the default Mixers and the Mixers on every card"""
         
-        for d in ['default'] + list(range(len(alsaaudio.cards()))):
-            if type(d) == type(0):
-                kwargs = { 'cardindex': d }
-            else:
-                kwargs = { 'device': d }
-
-            mixers = alsaaudio.mixers(**kwargs)
+        for c in alsaaudio.card_indexes():
+            mixers = alsaaudio.mixers(cardindex=c)
             
             for m in mixers:
-                mixer = alsaaudio.Mixer(m, **kwargs)
+                mixer = alsaaudio.Mixer(m, cardindex=c)
                 mixer.close()
 
     def testMixerAll(self):
@@ -90,14 +85,10 @@ class PCMTest(unittest.TestCase):
     """Test PCM objects"""
 
     def testPCM(self):
-        "Open a PCM object on every device"
+        "Open a PCM object on every card"
 
-        for pd in alsaaudio.pcms():
-            pcm = alsaaudio.PCM(device=pd)
-            pcm.close()
-
-        for pd in alsaaudio.pcms(alsaaudio.PCM_CAPTURE):
-            pcm = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, device=pd)
+        for c in alsaaudio.card_indexes():
+            pcm = alsaaudio.PCM(cardindex=c)
             pcm.close()
 
     def testPCMAll(self):