auracaster/sw_pyalsaaudio
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Decided that getenum should return the selected item and the available

Browse Source 
items.

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

Smallish documentation improvements.


git-svn-id: svn://svn.code.sf.net/p/pyalsaaudio/code/trunk@23 ec2f30ec-7544-0410-870e-f70ca00c83f0
This commit is contained in:
larsimmisch
2008-05-21 14:06:26 +00:00
parent 40c4386803
commit df89c12581
19 changed files with 585 additions and 354 deletions
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4
CHANGES
View File

@@ -1,3 +1,7 @@
Version 0.4:
- added mixer.getenum()
- small documentation improvements
Version 0.3:
- wrapped blocking calls with Py_BEGIN_ALLOW_THREADS/Py_END_ALLOW_THREADS
- added pause

4
README
View File

@@ -1,7 +1,7 @@
PyAlsaAudio
===========
Author: Casper Wilstrup (cwi@unispeed.dk)
Author: Casper Wilstrup (cwi@aves.dk)
This package contains wrappers for accessing the ALSA api from Python. It
is currently fairly complete for PCM devices. My next goal is to have
@@ -52,4 +52,4 @@ stdin
'recordtest.py' which captures sound from the microphone at writes
it raw to stdout.
'mixertest.py' which can be used to manipulate the mixers
'mixertest.py' which can be used to manipulate the mixers.

250
alsaaudio.c
View File

@@ -191,7 +191,7 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
snd_pcm_hw_params_current(self->handle,hwparams);
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":dumpinfo")) return NULL;
printf("PCM handle name = '%s'\n", snd_pcm_name(self->handle));
printf("PCM state = %s\n", snd_pcm_state_name(snd_pcm_state(self->handle)));
@@ -276,7 +276,7 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
static PyObject *
alsapcm_pcmtype(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":pcmtype")) return NULL;
return PyInt_FromLong(self->pcmtype);
}
@@ -288,7 +288,7 @@ Returns either PCM_CAPTURE or PCM_PLAYBACK.");
static PyObject *
alsapcm_pcmmode(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,"pcmmode")) return NULL;
return PyInt_FromLong(self->pcmmode);
}
@@ -303,7 +303,7 @@ Returns the mode of the PCM object. One of:\n\
static PyObject *
alsapcm_cardname(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":cardname")) return NULL;
return PyString_FromString(self->cardname);
}
@@ -317,7 +317,7 @@ static PyObject *
alsapcm_setchannels(alsapcm_t *self, PyObject *args) {
int channels;
int res;
if (!PyArg_ParseTuple(args,"i",&channels)) return NULL;
if (!PyArg_ParseTuple(args,"i:setchannels",&channels)) return NULL;
self->channels = channels;
res = alsapcm_setup(self);
if (res < 0) {
@@ -340,7 +340,7 @@ static PyObject *
alsapcm_setrate(alsapcm_t *self, PyObject *args) {
int rate;
int res;
if (!PyArg_ParseTuple(args,"i",&rate)) return NULL;
if (!PyArg_ParseTuple(args,"i:setrate",&rate)) return NULL;
self->rate = rate;
res = alsapcm_setup(self);
if (res < 0) {
@@ -361,7 +361,7 @@ static PyObject *
alsapcm_setformat(alsapcm_t *self, PyObject *args) {
int format;
int res;
if (!PyArg_ParseTuple(args,"i",&format)) return NULL;
if (!PyArg_ParseTuple(args,"i:setformat",&format)) return NULL;
self->format = format;
res = alsapcm_setup(self);
if (res < 0) {
@@ -379,7 +379,7 @@ static PyObject *
alsapcm_setperiodsize(alsapcm_t *self, PyObject *args) {
int periodsize;
int res;
if (!PyArg_ParseTuple(args,"i",&periodsize)) return NULL;
if (!PyArg_ParseTuple(args,"i:setperiodsize",&periodsize)) return NULL;
self->periodsize = periodsize;
res = alsapcm_setup(self);
if (res < 0) {
@@ -408,7 +408,7 @@ alsapcm_read(alsapcm_t *self, PyObject *args) {
return NULL;
}
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":read")) return NULL;
if (self->pcmtype != SND_PCM_STREAM_CAPTURE) {
PyErr_SetString(ALSAAudioError,"Cannot read from playback PCM");
return NULL;
@@ -453,7 +453,7 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) {
char *data;
int datalen;
int res;
if (!PyArg_ParseTuple(args,"s#",&data,&datalen)) return NULL;
if (!PyArg_ParseTuple(args,"s#:write",&data,&datalen)) return NULL;
if (datalen%self->framesize) {
PyErr_SetString(ALSAAudioError,
"Data size must be a multiple of framesize");
@@ -500,7 +500,7 @@ written at a later time.");
static PyObject *alsapcm_pause(alsapcm_t *self, PyObject *args) {
int enabled=1, res;
if (!PyArg_ParseTuple(args,"|i",&enabled)) return NULL;
if (!PyArg_ParseTuple(args,"|i:pause",&enabled)) return NULL;
Py_BEGIN_ALLOW_THREADS
res = snd_pcm_pause(self->handle, enabled);
@@ -632,7 +632,7 @@ alsamixer_list(PyObject *self, PyObject *args) {
char *cardname = "default";
PyObject *result = PyList_New(0);
if (!PyArg_ParseTuple(args,"|s",&cardname)) return NULL;
if (!PyArg_ParseTuple(args,"|s:mixers",&cardname)) return NULL;
snd_mixer_selem_id_alloca(&sid);
err = alsamixer_gethandle(cardname,&handle);
@@ -751,8 +751,9 @@ alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
if (snd_mixer_selem_is_playback_mono(elem)) self->pchannels = 1;
else {
for (channel=0; channel <= SND_MIXER_SCHN_LAST; channel++) {
if (snd_mixer_selem_has_playback_channel(elem, channel)) self->pchannels++;
else break;
if (snd_mixer_selem_has_playback_channel(elem, channel))
self->pchannels++;
else break;
}
}
}
@@ -762,8 +763,9 @@ alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
if (snd_mixer_selem_is_capture_mono(elem)) self->cchannels = 1;
else {
for (channel=0; channel <= SND_MIXER_SCHN_LAST; channel++) {
if (snd_mixer_selem_has_capture_channel(elem, channel)) self->cchannels++;
else break;
if (snd_mixer_selem_has_capture_channel(elem, channel))
self->cchannels++;
else break;
}
}
}
@@ -784,7 +786,7 @@ static void alsamixer_dealloc(alsamixer_t *self) {
static PyObject *
alsamixer_cardname(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":cardname")) return NULL;
return PyString_FromString(self->cardname);
}
@@ -796,7 +798,7 @@ 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,"")) return NULL;
if (!PyArg_ParseTuple(args,":mixer")) return NULL;
return PyString_FromString(self->controlname);
}
@@ -809,7 +811,7 @@ for example 'Master' or 'PCM'");
static PyObject *
alsamixer_mixerid(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":mixerid")) return NULL;
return PyInt_FromLong(self->controlid);
}
@@ -822,7 +824,7 @@ Returns the ID of the ALSA mixer controlled by this object.");
static PyObject *
alsamixer_volumecap(alsamixer_t *self, PyObject *args) {
PyObject *result;
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":volumecap")) return NULL;
result = PyList_New(0);
if (self->volume_cap&MIXER_CAP_VOLUME)
PyList_Append(result,PyString_FromString("Volume"));
@@ -856,7 +858,7 @@ Possible values in this list are:\n\
static PyObject *
alsamixer_switchcap(alsamixer_t *self, PyObject *args) {
PyObject *result;
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":switchcap")) return NULL;
result = PyList_New(0);
if (self->volume_cap&MIXER_CAP_SWITCH)
PyList_Append(result,PyString_FromString("Mute"));
@@ -920,7 +922,7 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args) {
char *dirstr = 0;
PyObject *result;
if (!PyArg_ParseTuple(args,"|s",&dirstr)) return NULL;
if (!PyArg_ParseTuple(args,"|s:getvolume",&dirstr)) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
@@ -948,8 +950,8 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args) {
&& snd_mixer_selem_has_capture_volume(elem)) {
snd_mixer_selem_get_capture_volume(elem, channel, &ival);
PyList_Append(
result,PyInt_FromLong(alsamixer_getpercentage(self->cmin,
self->cmax, ival)));
result, PyInt_FromLong(alsamixer_getpercentage(self->cmin,
self->cmax, ival)));
}
}
return result;
@@ -972,9 +974,8 @@ alsamixer_getrange(alsamixer_t *self, PyObject *args) {
snd_mixer_elem_t *elem;
int direction;
char *dirstr = 0;
PyObject *result;
if (!PyArg_ParseTuple(args,"|s",&dirstr)) return NULL;
if (!PyArg_ParseTuple(args,"|s:getrange",&dirstr)) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
@@ -985,20 +986,31 @@ alsamixer_getrange(alsamixer_t *self, PyObject *args) {
else if (strcasecmp(dirstr,"playback")==0) direction = 0;
else if (strcasecmp(dirstr,"capture")==0) direction = 1;
else {
PyErr_SetString(ALSAAudioError,"Invalid direction argument for direction");
PyErr_SetString(ALSAAudioError,"Invalid argument for direction");
return NULL;
}
result = PyList_New(0);
if (direction == 0 && snd_mixer_selem_has_playback_channel(elem, 0)) {
PyList_Append(result,PyInt_FromLong(self->pmin));
PyList_Append(result,PyInt_FromLong(self->pmax));
if (direction == 0) {
if (snd_mixer_selem_has_playback_channel(elem, 0)) {
return Py_BuildValue("[ii]", self->pmin, self->pmax);
}
else if (direction == 1 && snd_mixer_selem_has_capture_channel(elem, 0)
&& snd_mixer_selem_has_capture_volume(elem)) {
PyList_Append(result,PyInt_FromLong(self->cmin));
PyList_Append(result,PyInt_FromLong(self->cmax));
PyErr_SetString(ALSAAudioError, "Mixer has no playback channel");
return NULL;
}
else if (direction == 1) {
if (snd_mixer_selem_has_capture_channel(elem, 0)
&& snd_mixer_selem_has_capture_volume(elem)) {
return Py_BuildValue("[ii]", self->cmin, self->cmax);
}
return result;
PyErr_SetString(ALSAAudioError, "Mixer has no capture channel "
"or capture volume");
return NULL;
}
// Unreached statement
PyErr_SetString(ALSAAudioError,"Huh?");
return NULL;
}
PyDoc_STRVAR(getrange_doc,
@@ -1015,40 +1027,73 @@ if the mixer has this capability, otherwise 'capture'");
static PyObject *
alsamixer_getenum(alsamixer_t *self, PyObject *args) {
snd_mixer_elem_t *elem;
PyObject *elems;
int i, count, rc;
unsigned int index;
char name[32];
int res;
PyObject *result;
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args, ":getenum")) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!snd_mixer_selem_is_enumerated(elem)) {
PyErr_SetString(ALSAAudioError,"Mixer is no enumerated control");
return NULL;
}
// Not an enumerated control, return an empty tuple
return PyTuple_New(0);
}
res=snd_mixer_selem_get_enum_item(elem, 0, &index);
if(res) {
PyErr_SetString(ALSAAudioError, snd_strerror(res));
count = snd_mixer_selem_get_enum_items(elem);
if (count < 0) {
PyErr_SetString(ALSAAudioError, snd_strerror(count));
return NULL;
}
res=snd_mixer_selem_get_enum_item_name(elem, index, sizeof(name)-1, name);
if(res) {
PyErr_SetString(ALSAAudioError, snd_strerror(res));
result = PyTuple_New(2);
if (!result)
return NULL;
rc = snd_mixer_selem_get_enum_item(elem, 0, &index);
if(rc) {
PyErr_SetString(ALSAAudioError, snd_strerror(rc));
return NULL;
} else {
result = PyList_New(0);
PyList_Append(result,PyString_FromString(name));
}
rc = snd_mixer_selem_get_enum_item_name(elem, index, sizeof(name)-1, name);
if (rc) {
Py_DECREF(result);
PyErr_SetString(ALSAAudioError, snd_strerror(rc));
return NULL;
}
PyTuple_SetItem(result, 0, PyString_FromString(name));
elems = PyList_New(count);
if (!elems)
{
Py_DECREF(result);
return NULL;
}
for (i = 0; i < count; ++i) {
rc = snd_mixer_selem_get_enum_item_name(elem, i, sizeof(name)-1, name);
if (rc) {
Py_DECREF(elems);
Py_DECREF(result);
PyErr_SetString(ALSAAudioError, snd_strerror(rc));
return NULL;
}
PyList_SetItem(elems, i, PyString_FromString(name));
}
PyTuple_SetItem(result, 1, elems);
return result;
}
PyDoc_STRVAR(getenum_doc,
"getenum([direction]) -> List of enumerated controls (string)\n\
"getenum() -> Tuple of (string, list of strings)\n\
\n\
Returns a list of strings with the enumerated controls.");
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_getmute(alsamixer_t *self, PyObject *args) {
@@ -1056,7 +1101,7 @@ alsamixer_getmute(alsamixer_t *self, PyObject *args) {
int i;
int ival;
PyObject *result;
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":getmute")) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!snd_mixer_selem_has_playback_switch(elem)) {
@@ -1088,7 +1133,7 @@ alsamixer_getrec(alsamixer_t *self, PyObject *args) {
int i;
int ival;
PyObject *result;
if (!PyArg_ParseTuple(args,"")) return NULL;
if (!PyArg_ParseTuple(args,":getrec")) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!snd_mixer_selem_has_capture_switch(elem)) {
@@ -1124,7 +1169,9 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args) {
int channel = MIXER_CHANNEL_ALL;
int done = 0;
if (!PyArg_ParseTuple(args,"l|is",&volume,&channel,&dirstr)) return NULL;
if (!PyArg_ParseTuple(args,"l|is:setvolume",&volume,&channel,&dirstr))
return NULL;
if (volume < 0 || volume > 100) {
PyErr_SetString(ALSAAudioError,"Volume must be between 0 and 100");
return NULL;
@@ -1146,16 +1193,16 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args) {
for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) {
if (channel == -1 || channel == i) {
if (direction == 0 && 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);
done++;
physvolume = alsamixer_getphysvolume(self->pmin,self->pmax,volume);
snd_mixer_selem_set_playback_volume(elem, i, physvolume);
done++;
}
else if (direction == 1
&& snd_mixer_selem_has_capture_channel(elem, channel)
&& snd_mixer_selem_has_capture_volume(elem)) {
physvolume = alsamixer_getphysvolume(self->cmin,self->cmax,volume);
snd_mixer_selem_set_capture_volume(elem, i, physvolume);
done++;
physvolume = alsamixer_getphysvolume(self->cmin,self->cmax,volume);
snd_mixer_selem_set_capture_volume(elem, i, physvolume);
done++;
}
}
}
@@ -1190,7 +1237,7 @@ alsamixer_setmute(alsamixer_t *self, PyObject *args) {
int mute = 0;
int done = 0;
int channel = MIXER_CHANNEL_ALL;
if (!PyArg_ParseTuple(args,"i|i",&mute,&channel)) return NULL;
if (!PyArg_ParseTuple(args,"i|i:setmute",&mute,&channel)) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!snd_mixer_selem_has_playback_switch(elem)) {
@@ -1200,8 +1247,8 @@ alsamixer_setmute(alsamixer_t *self, PyObject *args) {
for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) {
if (channel == MIXER_CHANNEL_ALL || channel == i) {
if (snd_mixer_selem_has_playback_channel(elem, i)) {
snd_mixer_selem_set_playback_switch(elem, i, !mute);
done++;
snd_mixer_selem_set_playback_switch(elem, i, !mute);
done++;
}
}
}
@@ -1231,7 +1278,7 @@ alsamixer_setrec(alsamixer_t *self, PyObject *args) {
int rec = 0;
int done = 0;
int channel = MIXER_CHANNEL_ALL;
if (!PyArg_ParseTuple(args,"i|i",&rec,&channel)) return NULL;
if (!PyArg_ParseTuple(args,"i|i:setrec",&rec,&channel)) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!snd_mixer_selem_has_capture_switch(elem)) {
@@ -1369,39 +1416,52 @@ void initalsaaudio(void) {
}
_EXPORT_INT(m,"PCM_PLAYBACK",SND_PCM_STREAM_PLAYBACK);
_EXPORT_INT(m,"PCM_CAPTURE",SND_PCM_STREAM_CAPTURE);
_EXPORT_INT(m, "PCM_PLAYBACK",SND_PCM_STREAM_PLAYBACK);
_EXPORT_INT(m, "PCM_CAPTURE",SND_PCM_STREAM_CAPTURE);
_EXPORT_INT(m,"PCM_NORMAL",0);
_EXPORT_INT(m,"PCM_NONBLOCK",SND_PCM_NONBLOCK);
_EXPORT_INT(m,"PCM_ASYNC",SND_PCM_ASYNC);
_EXPORT_INT(m, "PCM_NORMAL",0);
_EXPORT_INT(m, "PCM_NONBLOCK",SND_PCM_NONBLOCK);
_EXPORT_INT(m, "PCM_ASYNC",SND_PCM_ASYNC);
/* PCM Formats */
_EXPORT_INT(m,"PCM_FORMAT_S8",SND_PCM_FORMAT_S8);
_EXPORT_INT(m,"PCM_FORMAT_U8",SND_PCM_FORMAT_U8);
_EXPORT_INT(m,"PCM_FORMAT_S16_LE",SND_PCM_FORMAT_S16_LE);
_EXPORT_INT(m,"PCM_FORMAT_S16_BE",SND_PCM_FORMAT_S16_BE);
_EXPORT_INT(m,"PCM_FORMAT_U16_LE",SND_PCM_FORMAT_U16_LE);
_EXPORT_INT(m,"PCM_FORMAT_U16_BE",SND_PCM_FORMAT_U16_BE);
_EXPORT_INT(m,"PCM_FORMAT_S24_LE",SND_PCM_FORMAT_S24_LE);
_EXPORT_INT(m,"PCM_FORMAT_S24_BE",SND_PCM_FORMAT_S24_BE);
_EXPORT_INT(m,"PCM_FORMAT_U24_LE",SND_PCM_FORMAT_U24_LE);
_EXPORT_INT(m,"PCM_FORMAT_U24_BE",SND_PCM_FORMAT_U24_BE);
_EXPORT_INT(m,"PCM_FORMAT_S32_LE",SND_PCM_FORMAT_S32_LE);
_EXPORT_INT(m,"PCM_FORMAT_S32_BE",SND_PCM_FORMAT_S32_BE);
_EXPORT_INT(m,"PCM_FORMAT_U32_LE",SND_PCM_FORMAT_U32_LE);
_EXPORT_INT(m,"PCM_FORMAT_U32_BE",SND_PCM_FORMAT_U32_BE);
_EXPORT_INT(m,"PCM_FORMAT_FLOAT_LE",SND_PCM_FORMAT_FLOAT_LE);
_EXPORT_INT(m,"PCM_FORMAT_FLOAT_BE",SND_PCM_FORMAT_FLOAT_BE);
_EXPORT_INT(m,"PCM_FORMAT_FLOAT64_LE",SND_PCM_FORMAT_FLOAT64_LE);
_EXPORT_INT(m,"PCM_FORMAT_FLOAT64_BE",SND_PCM_FORMAT_FLOAT64_BE);
_EXPORT_INT(m,"PCM_FORMAT_MU_LAW",SND_PCM_FORMAT_MU_LAW);
_EXPORT_INT(m,"PCM_FORMAT_A_LAW",SND_PCM_FORMAT_A_LAW);
_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_S8",SND_PCM_FORMAT_S8);
_EXPORT_INT(m, "PCM_FORMAT_U8",SND_PCM_FORMAT_U8);
_EXPORT_INT(m, "PCM_FORMAT_S16_LE",SND_PCM_FORMAT_S16_LE);
_EXPORT_INT(m, "PCM_FORMAT_S16_BE",SND_PCM_FORMAT_S16_BE);
_EXPORT_INT(m, "PCM_FORMAT_U16_LE",SND_PCM_FORMAT_U16_LE);
_EXPORT_INT(m, "PCM_FORMAT_U16_BE",SND_PCM_FORMAT_U16_BE);
_EXPORT_INT(m, "PCM_FORMAT_S24_LE",SND_PCM_FORMAT_S24_LE);
_EXPORT_INT(m, "PCM_FORMAT_S24_BE",SND_PCM_FORMAT_S24_BE);
_EXPORT_INT(m, "PCM_FORMAT_U24_LE",SND_PCM_FORMAT_U24_LE);
_EXPORT_INT(m, "PCM_FORMAT_U24_BE",SND_PCM_FORMAT_U24_BE);
_EXPORT_INT(m, "PCM_FORMAT_S32_LE",SND_PCM_FORMAT_S32_LE);
_EXPORT_INT(m, "PCM_FORMAT_S32_BE",SND_PCM_FORMAT_S32_BE);
_EXPORT_INT(m, "PCM_FORMAT_U32_LE",SND_PCM_FORMAT_U32_LE);
_EXPORT_INT(m, "PCM_FORMAT_U32_BE",SND_PCM_FORMAT_U32_BE);
_EXPORT_INT(m, "PCM_FORMAT_FLOAT_LE",SND_PCM_FORMAT_FLOAT_LE);
_EXPORT_INT(m, "PCM_FORMAT_FLOAT_BE",SND_PCM_FORMAT_FLOAT_BE);
_EXPORT_INT(m, "PCM_FORMAT_FLOAT64_LE",SND_PCM_FORMAT_FLOAT64_LE);
_EXPORT_INT(m, "PCM_FORMAT_FLOAT64_BE",SND_PCM_FORMAT_FLOAT64_BE);
_EXPORT_INT(m, "PCM_FORMAT_MU_LAW",SND_PCM_FORMAT_MU_LAW);
_EXPORT_INT(m, "PCM_FORMAT_A_LAW",SND_PCM_FORMAT_A_LAW);
_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);
/* Mixer stuff */
_EXPORT_INT(m,"MIXER_CHANNEL_ALL",MIXER_CHANNEL_ALL);
_EXPORT_INT(m, "MIXER_CHANNEL_ALL", MIXER_CHANNEL_ALL);
#if 0 // Omit for now - use case unknown
_EXPORT_INT(m, "MIXER_SCHN_UNKNOWN", SND_MIXER_SCHN_UNKNOWN);
_EXPORT_INT(m, "MIXER_SCHN_FRONT_LEFT", SND_MIXER_SCHN_FRONT_LEFT);
_EXPORT_INT(m, "MIXER_SCHN_FRONT_RIGHT", SND_MIXER_SCHN_FRONT_RIGHT);
_EXPORT_INT(m, "MIXER_SCHN_REAR_LEFT", SND_MIXER_SCHN_REAR_LEFT);
_EXPORT_INT(m, "MIXER_SCHN_REAR_RIGHT", SND_MIXER_SCHN_REAR_RIGHT);
_EXPORT_INT(m, "MIXER_SCHN_FRONT_CENTER", SND_MIXER_SCHN_FRONT_CENTER);
_EXPORT_INT(m, "MIXER_SCHN_WOOFER", SND_MIXER_SCHN_WOOFER);
_EXPORT_INT(m, "MIXER_SCHN_SIDE_LEFT", SND_MIXER_SCHN_SIDE_LEFT);
_EXPORT_INT(m, "MIXER_SCHN_SIDE_RIGHT", SND_MIXER_SCHN_SIDE_RIGHT);
_EXPORT_INT(m, "MIXER_SCHN_REAR_CENTER", SND_MIXER_SCHN_REAR_CENTER);
_EXPORT_INT(m, "MIXER_SCHN_MONO", SND_MIXER_SCHN_MONO);
#endif
}

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@@ -103,7 +103,7 @@ About this document ...</a>
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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2
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@@ -99,7 +99,7 @@ Contents</a>
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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17
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@@ -65,15 +65,16 @@ whatsoever.
<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.
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.
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>
@@ -112,7 +113,7 @@ me.
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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3
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@@ -44,6 +44,7 @@
<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>
@@ -109,7 +110,7 @@
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->

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@@ -60,19 +60,23 @@ Mixer objects provides access to the ALSA mixer API.
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><span class="typelabel">class</span>&nbsp;<tt id='l2h-16' xml:id='l2h-16' class="class">Mixer</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>control</var><big>]</big><var>, </var><big>[</big><var>id</var><big>]</big><var>, </var><big>[</big><var>cardname</var><big>]</big><var></var>)</td></tr></table></dt>
<td><nobr><b><span class="typelabel">class</span>&nbsp;<tt id='l2h-17' xml:id='l2h-17' class="class">Mixer</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>control</var><big>]</big><var>, </var><big>[</big><var>id</var><big>]</big><var>,
</var><big>[</big><var>cardname</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
<var>control</var> - specifies which control to manipulate using this mixer object. The list
of available controls can be found with the <tt class="module">alsaaudio</tt>.<tt class="function">mixers</tt> function.
The default value is 'Master' - other common controls include 'Master Mono', 'PCM', 'Line', etc.
<var>control</var> - specifies which control to manipulate using this
mixer object. The list of available controls can be found with the
<tt class="module">alsaaudio</tt>.<tt class="function">mixers</tt> function. The default value is
'Master' - other common controls include 'Master Mono', 'PCM',
'Line', etc.
<p>
<var>id</var> - the id of the mixer control. Default is 0
<p>
<var>cardname</var> - specifies which card should be used (this is only relevant
if you have more than one sound card). Omit to use the default sound card
<var>cardname</var> - specifies which card should be used (this is only
relevant if you have more than one sound card). Omit to use the
default sound card
</dl>
<p>
@@ -80,36 +84,36 @@ Mixer objects have the following methods:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-17' xml:id='l2h-17' class="method">cardname</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-18' xml:id='l2h-18' class="method">cardname</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the name of the sound card used by this Mixer object
Return the name of the sound card used by this Mixer object
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-18' xml:id='l2h-18' class="method">mixer</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-19' xml:id='l2h-19' class="method">mixer</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the name of the specific mixer controlled by this object, For example 'Master'
or 'PCM'
Return the name of the specific mixer controlled by this object, For
example 'Master' or 'PCM'
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-19' xml:id='l2h-19' class="method">mixerid</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-20' xml:id='l2h-20' class="method">mixerid</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the ID of the ALSA mixer controlled by this object.
Return the ID of the ALSA mixer controlled by this object.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-20' xml:id='l2h-20' class="method">switchcap</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-21' xml:id='l2h-21' class="method">switchcap</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Returns a list of the switches which are defined by this specific mixer. Possible values in
this list are:
Returns a list of the switches which are defined by this specific
mixer. Possible values in this list are:
<p>
<div class="center"><table class="realtable">
@@ -137,16 +141,17 @@ this list are:
</table></div>
<p>
To manipulate these swithes use the <tt class="method">setrec</tt> or <tt class="method">setmute</tt> methods
To manipulate these swithes use the <tt class="method">setrec</tt> or
<tt class="method">setmute</tt> methods
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-21' xml:id='l2h-21' class="method">volumecap</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-22' xml:id='l2h-22' class="method">volumecap</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Returns a list of the volume control capabilities of this mixer. Possible values in
the list are:
Returns a list of the volume control capabilities of this mixer.
Possible values in the list are:
<p>
<div class="center"><table class="realtable">
@@ -160,15 +165,18 @@ the list are:
<tr><td class="left" valign="baseline"><volume capabilities>'Volume'</volume></td>
<td class="left" >This mixer can control volume</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Joined Volume'</volume></td>
<td class="left" >This mixer can control volume for all channels at the same time</td></tr>
<td class="left" >This mixer can control volume for all channels at
the same time</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Playback Volume'</volume></td>
<td class="left" >This mixer can manipulate the playback volume</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Joined Playback Volume'</volume></td>
<td class="left" >Manipulate playback volumne for all channels at the same time</td></tr>
<td class="left" >Manipulate playback volumne for all
channels at the same time</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Capture Volume'</volume></td>
<td class="left" >Manipulate sound capture volume</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Joined Capture Volume'</volume></td>
<td class="left" >Manipulate sound capture volume for all channels at a time</td></tr></tbody>
<td class="left" >Manipulate sound capture volume for all
channels at a time</td></tr></tbody>
</table></div>
<p>
@@ -176,34 +184,40 @@ the list are:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-22' xml:id='l2h-22' class="method">getrange</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<td><nobr><b><tt id='l2h-23' xml:id='l2h-23' class="method">getenum</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the volume range of the ALSA mixer controlled by this object.
For enumerated controls, return the currently selected item and
the list of items available.
<p>
The optional <var>direction</var> 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'
Returns a tuple <i>(string, list of strings)</i>.
<p>
</dl>
For example, my soundcard has a Mixer called <i>Mono Output Select</i>.
Using <i>amixer</i>, I get:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-23' xml:id='l2h-23' class="method">getvolume</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Returns a list with the current volume settings for each channel. The list elements
are integer percentages.
<div class="verbatim"><pre>
$ amixer get "Mono Output Select"
Simple mixer control 'Mono Output Select',0
Capabilities: enum
Items: 'Mix' 'Mic'
Item0: 'Mix'
</pre></div>
<p>
The optional <var>direction</var> 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'
Using <tt class="module">alsaaudio</tt>, one could do:
<div class="verbatim"><pre>
&gt;&gt;&gt; import alsaaudio
&gt;&gt;&gt; m = alsaaudio.Mixer('Mono Output Select')
&gt;&gt;&gt; m.getenum()
('Mix', ['Mix', 'Mic'])
</pre></div>
<p>
This method will return an empty tuple if the mixer is not an
enumerated control.
</dl>
<p>
@@ -211,52 +225,95 @@ if the mixer has this capability, otherwise 'capture'
<td><nobr><b><tt id='l2h-24' xml:id='l2h-24' class="method">getmute</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted.
Return a list indicating the current mute setting for each channel.
0 means not muted, 1 means muted.
<p>
This method will fail if the mixer has no playback switch capabilities.
This method will fail if the mixer has no playback switch
capabilities.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-25' xml:id='l2h-25' class="method">getrec</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-25' xml:id='l2h-25' class="method">getrange</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Return the volume range of the ALSA mixer controlled by this object.
<p>
The optional <var>direction</var> 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'
<p>
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-26' xml:id='l2h-26' class="method">getrec</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return a list indicating the current record mute setting for each channel. 0 means not recording, 1
means not recording.
Return a list indicating the current record mute setting for each
channel. 0 means not recording, 1 means recording.
<p>
This method will fail if the mixer has no capture switch capabilities.
This method will fail if the mixer has no capture switch
capabilities.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-26' xml:id='l2h-26' class="method">setvolume</tt></b>(</nobr></td>
<td><var>volume,</var><big>[</big><var>channel</var><big>]</big><var>,</var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<td><nobr><b><tt id='l2h-27' xml:id='l2h-27' class="method">getvolume</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Change the current volume settings for this mixer. The <var>volume</var> argument controls
the new volume setting as an integer percentage.
Returns a list with the current volume settings for each channel.
The list elements are integer percentages.
<p>
If the optional argument <var>channel</var> 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</var> 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'
<p>
The optional <var>direction</var> 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'.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-27' xml:id='l2h-27' class="method">setmute</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-28' xml:id='l2h-28' class="method">setvolume</tt></b>(</nobr></td>
<td><var>volume,</var><big>[</big><var>channel</var><big>]</big><var>,
</var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
<p>
Change the current volume settings for this mixer. The <var>volume</var>
argument controls the new volume setting as an integer percentage.
<p>
If the optional argument <var>channel</var> is present, the volume is set
only for this channel. This assumes that the mixer can control the
volume for the channels independently.
<p>
The optional <var>direction</var> 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'.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-29' xml:id='l2h-29' class="method">setmute</tt></b>(</nobr></td>
<td><var>mute, </var><big>[</big><var>channel</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Sets the mute flag to a new value. The <var>mute</var> argument is either 0 for not muted, or 1 for muted.
Sets the mute flag to a new value. The <var>mute</var> argument is either
0 for not muted, or 1 for muted.
<p>
The optional <var>channel</var> argument controls which channel is muted. The default is to set the mute flag
for all channels.
The optional <var>channel</var> argument controls which channel is muted.
The default is to set the mute flag for all channels.
<p>
This method will fail if the mixer has no playback mute capabilities
@@ -264,42 +321,48 @@ This method will fail if the mixer has no playback mute capabilities
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-28' xml:id='l2h-28' class="method">setrec</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-30' xml:id='l2h-30' class="method">setrec</tt></b>(</nobr></td>
<td><var>capture,</var><big>[</big><var>channel</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Sets the capture mute flag to a new value. The <var>capture</var> argument is either 0 for no capture,
or 1 for capture.
Sets the capture mute flag to a new value. The <var>capture</var>
argument is either 0 for no capture, or 1 for capture.
<p>
The optional <var>channel</var> argument controls which channel is changed. The default is to set the capture flag
for all channels.
The optional <var>channel</var> argument controls which channel is
changed. The default is to set the capture flag for all channels.
<p>
This method will fail if the mixer has no capture switch capabilities
This method will fail if the mixer has no capture switch
capabilities.
</dl>
<p>
<b>A Note on the ALSA Mixer API</b>
<p>
The ALSA mixer API is extremely complicated - and hardly documented at all. <tt class="module">alsaaudio</tt> 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 <tt class="module">alsaaudio</tt> API is easy to understand and use.
The ALSA mixer API is extremely complicated - and hardly documented at
all. <tt class="module">alsaaudio</tt> 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 <tt class="module">alsaaudio</tt> API is easy to understand and use.
<p>
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 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.
<p>
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.
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.
<p>
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.
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.
<p>
@@ -336,7 +399,7 @@ and that which I do understand has come from a painful trial and error process.
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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@@ -60,8 +60,7 @@
class="platform">Linux</span>.</p>
<p>
tex2html_comment_mark>15
<p>
<p>
@@ -83,12 +82,14 @@ sound card). Omit to use the default sound card.
<td><nobr><b><span class="typelabel">class</span>&nbsp;<tt id='l2h-3' xml:id='l2h-3' class="class">PCM</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>type</var><big>]</big><var>, </var><big>[</big><var>mode</var><big>]</big><var>, </var><big>[</big><var>cardname</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
This class is used to represent a PCM device (both playback and capture devices).
The arguments are:
<br><var>type</var> - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
<br><var>mode</var> - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).
<br><var>cardname</var> - specifies which card should be used (this is only relevant
if you have more than one sound card). Omit to use the default sound card
This class is used to represent a PCM device (both playback and
capture devices).
The arguments are:
<br> <var>type</var> - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
<br> <var>mode</var> - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).
<br> <var>cardname</var> - specifies which card should be used (this is only
relevant if you have more than one sound card). Omit to use the
default sound card
</dl>
<p>
@@ -99,7 +100,7 @@ if you have more than one sound card). Omit to use the default sound card
This class is used to access a specific ALSA mixer.
The arguments are:
<br><var>control</var> - Name of the chosen mixed (default is Master).
<br><var>id</var> - id of mixer (default is 0) - More explaniation needed here
<br><var>id</var> - id of mixer (default is 0) - More explanation needed here
<br><var>cardname</var> specifies which card should be used (this is only relevant
if you have more than one sound card). Omit to use the default sound card
</dl>
@@ -107,9 +108,9 @@ if you have more than one sound card). Omit to use the default sound card
<p>
<dl><dt><b><span class="typelabel">exception</span>&nbsp;<tt id='l2h-5' xml:id='l2h-5' class="exception">ALSAAudioError</tt></b></dt>
<dd>
Exception raised when an operation fails for a ALSA specific reason.
The exception argument is a string describing the reason of the
failure.
Exception raised when an operation fails for a ALSA specific reason.
The exception argument is a string describing the reason of the
failure.
</dd></dl>
<p>
@@ -161,7 +162,7 @@ failure.
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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@@ -55,7 +55,7 @@
<p>
The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI
functionality to the Linux operating system.
functionality to the Linux operating system.
<p>
Logically ALSA consists of these components:
@@ -113,7 +113,7 @@ More information about ALSA may be found on the project homepage
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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@@ -54,27 +54,29 @@
</h1>
<p>
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).
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).
<p>
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.
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.
<p>
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 of yet.
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.
<p>
PyAlsaAudio hass full support for sound capture, playback of sound, as well as
the ALSA Mixer API.
PyAlsaAudio hass full support for sound capture, playback of sound, as
well as the ALSA Mixer API.
<p>
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
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
<div class="navigation">
<div class='online-navigation'>
@@ -109,7 +111,7 @@ appreciated
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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19
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@@ -55,11 +55,18 @@
<p>
Note: the wrappers link with the alsasound library (from the alsa-lib
package). Verify that this is installed by looking for /usr/lib/libasound.so
before building. 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.
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.
<p>
On Debian (and probably Ubuntu), make sure you have libasound2-dev installed.
<p>
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.
<p>
To install, execute the following:
@@ -108,7 +115,7 @@ And then as root:
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
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@@ -60,72 +60,86 @@ terminology.
<p>
<dl>
<dt><strong>Sample</strong></dt>
<dd>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.
<dd>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.
<p>
</dd>
<dt><strong>Frame</strong></dt>
<dd>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.
<dd>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.
<p>
</dd>
<dt><strong>Frame size</strong></dt>
<dd>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
<dd>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
<p>
</dd>
<dt><strong>Rate</strong></dt>
<dd>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.
<dd>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.
<p>
</dd>
<dt><strong>Data rate</strong></dt>
<dd>This is the number of bytes, which must be recorded or provided per second
at a certain frame size and rate.
<dd>This is the number of bytes, which must be recorded
or provided per second at a certain frame size and rate.
<p>
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
8000 * 1 * 1 = 8 kb/s
<p>
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)
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)
<p>
</dd>
<dt><strong>Period</strong></dt>
<dd>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.
<dd>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.
<p>
</dd>
<dt><strong>Period size</strong></dt>
<dd>This is the size of each period in Hz. <em>Not bytes, but Hz!.</em> In <tt class="module">alsaaudio</tt>
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.
<dd>This is the size of each period in Hz. <em>Not
bytes, but Hz!.</em> In <tt class="module">alsaaudio</tt> 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.
<p>
</dd>
</dl>
<p>
Once you understand these concepts, you will be ready to actually utilize PCM API. Read on.
Once you understand these concepts, you will be ready to use the PCM
API. Read on.
<p>
@@ -162,7 +176,7 @@ Once you understand these concepts, you will be ready to actually utilize PCM AP
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->

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@@ -54,8 +54,9 @@
</h2>
<p>
For now, the only examples available are the 'playbacktest.py' and the 'recordtest.py' programs included.
This will change in a future version.
For now, the only examples available are the 'playbacktest.py' and the
'recordtest.py' programs included. This will change in a future
version.
<p>
@@ -92,7 +93,7 @@ This will change in a future version.
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->

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@@ -56,13 +56,15 @@
</h2>
<p>
The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA
and many other places to handle playback and capture of sampled sound data.
The acronym PCM is short for Pulse Code Modulation and is the method
used in ALSA and many other places to handle playback and capture of
sampled sound data.
<p>
PCM objects in <tt class="module">alsaaudio</tt> are used to do exactly that, either play sample based
sound or capture sound from some input source (perhaps a microphone). The PCM object
constructor takes the following arguments:
PCM objects in <tt class="module">alsaaudio</tt> are used to do exactly that, either
play sample based sound or capture sound from some input source
(probably a microphone). The PCM object constructor takes the following
arguments:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
@@ -74,18 +76,21 @@ constructor takes the following arguments:
<var>type</var> - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
<p>
<var>mode</var> - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).
In PCM_NONBLOCK mode, calls to read will return immediately independent of wether
there is any actual data to read. Similarly, write calls will return immediately
without actually writing anything to the playout buffer if the buffer is full.
<var>mode</var> - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the
default). In PCM_NONBLOCK mode, calls to read will return immediately
independent of wether there is any actual data to read. Similarly,
write calls will return immediately without actually writing anything
to the playout buffer if the buffer is full.
<p>
In the current version of <tt class="module">alsaaudio</tt> PCM_ASYNC is useless, since it relies
on a callback procedure, which can't be specified from Python.
In the current version of <tt class="module">alsaaudio</tt> PCM_ASYNC is useless,
since it relies on a callback procedure, which can't be specified through
this API yet.
<p>
<var>cardname</var> - specifies which card should be used (this is only relevant
if you have more than one sound card). Omit to use the default sound card
<var>cardname</var> - specifies which card should be used (this is only
relevant if you have more than one sound card). Omit to use the
default sound card
<p>
This will construct a PCM object with default settings:
@@ -108,7 +113,7 @@ PCM objects have the following methods:
<td><nobr><b><tt id='l2h-7' xml:id='l2h-7' class="method">pcmtype</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
</dl>
<p>
@@ -116,7 +121,8 @@ Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK.
<td><nobr><b><tt id='l2h-8' xml:id='l2h-8' class="method">pcmmode</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL
Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC,
or PCM_NORMAL
</dl>
<p>
@@ -124,7 +130,7 @@ Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL
<td><nobr><b><tt id='l2h-9' xml:id='l2h-9' class="method">cardname</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return the name of the sound card used by this PCM object.
Return the name of the sound card used by this PCM object.
</dl>
<p>
@@ -132,8 +138,9 @@ Return the name of the sound card used by this PCM object.
<td><nobr><b><tt id='l2h-10' xml:id='l2h-10' class="method">setchannels</tt></b>(</nobr></td>
<td><var>nchannels</var>)</td></tr></table></dt>
<dd>
Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo,
and 6 = full 6 channel audio. Few sound cards support more than 2 channels
Used to set the number of capture or playback channels. Common
values are: 1 = mono, 2 = stereo, and 6 = full 6 channel audio. Few
sound cards support more than 2 channels
</dl>
<p>
@@ -141,17 +148,18 @@ and 6 = full 6 channel audio. Few sound cards support more than 2 channels
<td><nobr><b><tt id='l2h-11' xml:id='l2h-11' class="method">setrate</tt></b>(</nobr></td>
<td><var>rate</var>)</td></tr></table></dt>
<dd>
Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality),
and 96000
Set the sample rate in Hz for the device. Typical values are 8000
(poor sound), 16000, 44100 (cd quality), and 96000
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-12' xml:id='l2h-12' class="method">setformat</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<td><var>format</var>)</td></tr></table></dt>
<dd>
The sound format of the device. Sound format controls how the PCM device interpret data for playback,
and how data is encoded in captures.
The sound <var>format</var> of the device. Sound format controls how the PCM
device interpret data for playback, and how data is encoded in
captures.
<p>
The following formats are provided by ALSA:
@@ -168,47 +176,66 @@ The following formats are provided by ALSA:
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U8</formats></td>
<td class="left" >Signed 8 bit samples for each channel</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S16_LE</formats></td>
<td class="left" >Signed 16 bit samples for each channel (Little Endian byte order)</td></tr>
<td class="left" >Signed 16 bit samples for each channel
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S16_BE</formats></td>
<td class="left" >Signed 16 bit samples for each channel (Big Endian byte order)</td></tr>
<td class="left" >Signed 16
bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U16_LE</formats></td>
<td class="left" >Unsigned 16 bit samples for each channel (Little Endian byte order)</td></tr>
<td class="left" >Unsigned 16 bit samples for each channel
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U16_BE</formats></td>
<td class="left" >Unsigned 16 bit samples for each channel (Big Endian byte order)</td></tr>
<td class="left" >Unsigned 16
bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S24_LE</formats></td>
<td class="left" >Signed 24 bit samples for each channel (Little Endian byte order)</td></tr>
<td class="left" >Signed 24 bit samples for each channel
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S24_BE</formats></td>
<td class="left" >Signed 24 bit samples for each channel (Big Endian byte order)</td></tr>
<td class="left" >Signed 24
bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U24_LE</formats></td>
<td class="left" >Unsigned 24 bit samples for each channel (Little Endian byte order)</td></tr>
<td class="left" >Unsigned 24 bit samples for each channel
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U24_BE</formats></td>
<td class="left" >Unsigned 24 bit samples for each channel (Big Endian byte order)</td></tr>
<td class="left" >Unsigned 24
bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S32_LE</formats></td>
<td class="left" >Signed 32 bit samples for each channel (Little Endian byte order)</td></tr>
<td class="left" >Signed 32 bit samples for each channel
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S32_BE</formats></td>
<td class="left" >Signed 32 bit samples for each channel (Big Endian byte order)</td></tr>
<td class="left" >Signed 32
bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U32_LE</formats></td>
<td class="left" >Unsigned 32 bit samples for each channel (Little Endian byte order)</td></tr>
<td class="left" >Unsigned 32 bit samples for each channel
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_U32_BE</formats></td>
<td class="left" >Unsigned 32 bit samples for each channel (Big Endian byte order)</td></tr>
<td class="left" >Unsigned 32
bit samples for each channel (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_FLOAT_LE</formats></td>
<td class="left" >32 bit samples encoded as float. (Little Endian byte order)</td></tr>
<td class="left" >32 bit samples encoded as float.
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_FLOAT_BE</formats></td>
<td class="left" >32 bit samples encoded as float (Big Endian byte order)</td></tr>
<td class="left" >32 bit
samples encoded as float (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_FLOAT64_LE</formats></td>
<td class="left" >64 bit samples encoded as float. (Little Endian byte order)</td></tr>
<td class="left" >64 bit samples encoded as float.
(Little Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_FLOAT64_BE</formats></td>
<td class="left" >64 bit samples encoded as float. (Big Endian byte order)</td></tr>
<td class="left" >64 bit
samples encoded as float. (Big Endian byte order)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_MU_LAW</formats></td>
<td class="left" >A logarithmic encoding (used by Sun .au files)</td></tr>
<td class="left" >A logarithmic encoding (used by Sun .au
files)</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_A_LAW</formats></td>
<td class="left" >Another logarithmic encoding</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_IMA_ADPCM</formats></td>
<td class="left" >a 4:1 compressed format defined by the Interactive Multimedia Association</td></tr>
<td class="left" >a 4:1 compressed format defined by the
Interactive Multimedia Association</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_MPEG</formats></td>
<td class="left" >MPEG encoded audio?</td></tr>
<td class="left" >MPEG
encoded audio?</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_GSM</formats></td>
<td class="left" >9600 constant rate encoding well suitet for speech</td></tr></tbody>
<td class="left" >9600 bits/s constant rate encoding for speech</td></tr></tbody>
</table></div>
<p>
@@ -219,9 +246,10 @@ The following formats are provided by ALSA:
<td><nobr><b><tt id='l2h-13' xml:id='l2h-13' class="method">setperiodsize</tt></b>(</nobr></td>
<td><var>period</var>)</td></tr></table></dt>
<dd>
Sets the actual period size in frames. Each write should consist of exactly this number of frames, and
each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case
it may return nothing at all)
Sets the actual period size in frames. Each write should consist of
exactly this number of frames, and each read will return this number
of frames (unless the device is in PCM_NONBLOCK mode, in which case
it may return nothing at all)
</dl>
<p>
@@ -229,14 +257,16 @@ it may return nothing at all)
<td><nobr><b><tt id='l2h-14' xml:id='l2h-14' class="method">read</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a
tuple (length,data) where <em>length</em> is the size in bytes of the captured data, and <em>data</em>
is the captured sound frames as a string. The length of the returned data will be periodsize*framesize
bytes.
In PCM_NORMAL mode, this function blocks until a full period is
available, and then returns a tuple (length,data) where
<em>length</em> is the number of frames of captured data, and
<em>data</em> is the captured sound frames as a string. The length of
the returned data will be periodsize*framesize bytes.
<p>
In PCM_NONBLOCK mode, the call will not block, but will return <code>(0,'')</code> if no new period
has become available since the last call to read.
In PCM_NONBLOCK mode, the call will not block, but will return
<code>(0,'')</code> if no new period has become available since the last
call to read.
</dl>
<p>
@@ -244,50 +274,67 @@ has become available since the last call to read.
<td><nobr><b><tt id='l2h-15' xml:id='l2h-15' class="method">write</tt></b>(</nobr></td>
<td><var>data</var>)</td></tr></table></dt>
<dd>
Writes (plays) the sound in data. The length of data <em>must</em> be a multiple of the frame size, and
<em>should</em> be exactly the size of a period. If less than 'period size' frames are provided, the actual
playout will not happen until more data is written.
Writes (plays) the sound in data. The length of data <em>must</em> be
a multiple of the frame size, and <em>should</em> be exactly the size
of a period. If less than 'period size' frames are provided, the
actual playout will not happen until more data is written.
<p>
If the device is not in PCM_NONBLOCK mode, this call will block if the kernel buffer is full, and
until enough sound has been played to allow the sound data to be buffered. The call always returns
the size of the data provided
If the device is not in PCM_NONBLOCK mode, this call will block if
the kernel buffer is full, and until enough sound has been played to
allow the sound data to be buffered. The call always returns the
size of the data provided
<p>
In PCM_NONBLOCK mode, the call will return immediately, with a return value of zero, if the buffer is
full. In this case, the data should be written at a later time.
In PCM_NONBLOCK mode, the call will return immediately, with a
return value of zero, if the buffer is full. In this case, the data
should be written at a later time.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-16' xml:id='l2h-16' class="method">pause</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>enable=1</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
If <var>enable</var> is 1, playback or capture is paused. If <var>enable</var> is 0,
playback/capture is resumed.
</dl>
<p>
<strong>A few hints on using PCM devices for playback</strong>
<p>
The most common reason for problems with playback of PCM audio, is that the people don't properly understand
that writes to PCM devices must match <em>exactly</em> the data rate of the device.
The most common reason for problems with playback of PCM audio, is
that the people don't properly understand that writes to PCM devices
must match <em>exactly</em> the data rate of the device.
<p>
If too little data is written to the device, it will underrun, and ugly clicking sounds will occur. Conversely,
of too much data is written to the device, the write function will either block (PCM_NORMAL mode) or return zero
(PCM_NONBLOCK mode).
If too little data is written to the device, it will underrun, and
ugly clicking sounds will occur. Conversely, of too much data is
written to the device, the write function will either block
(PCM_NORMAL mode) or return zero (PCM_NONBLOCK mode).
<p>
If your program does nothing, but play sound, the easiest way is to put the device in PCM_NORMAL mode, and just
write as much data to the device as possible. This strategy can also be achieved by using a separate thread
with the sole task of playing out sound.
If your program does nothing, but play sound, the easiest way is to
put the device in PCM_NORMAL mode, and just write as much data to the
device as possible. This strategy can also be achieved by using a
separate thread with the sole task of playing out sound.
<p>
In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few
periods by calling write a couple of times, and then use some timer method to write one period size of data to
the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer
doesn't expire exactly on time.
In GUI programs, however, it may be a better strategy to setup the
device, preload the buffer with a few periods by calling write a
couple of times, and then use some timer method to write one period
size of data to the device every period. The purpose of the preloading
is to avoid underrun clicks if the used timer doesn't expire exactly
on time.
<p>
Also note, that most timer APIs that you can find for Python will cummulate time delays: If you set the timer
to expire after 1/10'th of a second, the actual timeout will happen slightly later, which will accumulate to
quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add
extra writes as nessecary.
Also note, that most timer APIs that you can find for Python will
cummulate time delays: If you set the timer to expire after 1/10'th of
a second, the actual timeout will happen slightly later, which will
accumulate to quite a lot after a few seconds. Hint: use time.time()
to check how much time has really passed, and add extra writes as
nessecary.
<p>
@@ -324,7 +371,7 @@ extra writes as nessecary.
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
</div>
<!--End of Navigation Panel-->

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@@ -44,6 +44,7 @@
<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>
@@ -102,7 +103,7 @@
</div>
</div>
<hr />
<span class="release-info">Release 0.3.</span>
<span class="release-info">Release 0.4.</span>
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61
doc/src/libalsaaudio.tex
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@@ -383,6 +383,43 @@ To manipulate these swithes use the \method{setrec} or
\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.
@@ -393,6 +430,14 @@ To manipulate these swithes use the \method{setrec} or
\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.
@@ -404,22 +449,6 @@ To manipulate these swithes use the \method{setrec} or
\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]{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]{setvolume}{volume,\optional{channel},
\optional{direction}}

2
doc/src/pyalsaaudio.tex
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@@ -2,7 +2,7 @@
\title{PyAlsaAudio}
\release{0.3}
\release{0.4}
% At minimum, give your name and an email address. You can include a
% snail-mail address if you like.

2
setup.py
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@@ -18,7 +18,7 @@ if version < '2.2.3':
setup(
name = 'pyalsaaudio',
version = '0.3',
version = '0.4',
description = 'ALSA bindings',
long_description = __doc__,
author = 'Casper Wilstrup',