auracaster/pyalsaaudio
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- Added close methods to PCM and Mixer

Browse Source 
- Avoid internal re-opening of PCM objects when parameters change.

- PCM objects can be opened by card id
- Mixer objects can be openend by card index
  (this is not consistent, but ALSA works this way)

- Adjusted and cleaned up documentation.



git-svn-id: svn://svn.code.sf.net/p/pyalsaaudio/code/trunk@26 ec2f30ec-7544-0410-870e-f70ca00c83f0
This commit is contained in:
larsimmisch
2008-11-28 00:43:12 +00:00
parent 5c2a006553
commit dfb89bde62
16 changed files with 770 additions and 1936 deletions
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alsaaudio.c
+367 -64
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@@ -16,7 +16,8 @@
#include "Python.h"
#if PY_MAJOR_VERSION < 3
#include "stringobject.h"
#define PyUnicode_FromString PyString_FromString
#define PyUnicode_AS_DATA PyString_AS_STRING
#define PyUnicode_Check PyString_Check
#endif
#include <alsa/asoundlib.h>
#include <stdio.h>
@@ -71,14 +72,98 @@ typedef struct {
} alsamixer_t;
static PyObject *ALSAAudioError;
/* Translate a card id to a ALSA cardname
Returns a newly allocated string.
*/
char *translate_cardname(char *name)
{
static char dflt[] = "default";
char *full = NULL;
if (!name || !strcmp(name, dflt))
return strdup(dflt);
// If we find a colon, we assume it is a real ALSA cardname
if (strchr(name, ':'))
return strdup(name);
full = malloc(strlen("default:CARD=") + strlen(name) + 1);
sprintf(full, "default:CARD=%s", name);
return full;
}
/* Translate a card index to a ALSA cardname
Returns a newly allocated string.
*/
char *translate_cardidx(int idx)
{
char name[32];
sprintf(name, "hw:%d", idx);
return strdup(name);
}
/******************************************/
/* PCM object wrapper */
/******************************************/
static PyTypeObject ALSAPCMType;
static PyObject *ALSAAudioError;
static PyObject *
alsacard_list(PyObject *self, PyObject *args) {
int rc;
int card = -1;
snd_ctl_card_info_t *info;
snd_ctl_t *handle;
PyObject *result = NULL;
if (!PyArg_ParseTuple(args,":cards"))
return NULL;
snd_ctl_card_info_alloca(&info);
result = PyList_New(0);
for (rc = snd_card_next(&card); !rc && (card >= 0);
rc = snd_card_next(&card))
{
char name[32];
int err;
/* One would be tempted to think that snd_card_get_name returns a name
that is actually meaningful for any further operations.
Not in ALSA land. Here we need the id, not the name */
sprintf(name, "hw:%d", card);
if ((err = snd_ctl_open(&handle, name, 0)) < 0) {
PyErr_SetString(ALSAAudioError,snd_strerror(err));
return NULL;
}
if ((err = snd_ctl_card_info(handle, info)) < 0) {
PyErr_SetString(ALSAAudioError,snd_strerror(err));
snd_ctl_close(handle);
Py_DECREF(result);
return NULL;
}
PyList_Append(result,
PyUnicode_FromString(snd_ctl_card_info_get_id(info)));
snd_ctl_close(handle);
}
return result;
}
PyDoc_STRVAR(cards_doc,
"cards()\n\
\n\
List the available card ids.");
static int alsapcm_setup(alsapcm_t *self) {
int res,dir;
@@ -87,18 +172,12 @@ static int alsapcm_setup(alsapcm_t *self) {
snd_pcm_uframes_t frames;
snd_pcm_hw_params_t *hwparams;
if (self->handle) {
snd_pcm_close(self->handle);
self->handle = 0;
}
res = snd_pcm_open(&(self->handle),self->cardname,self->pcmtype,
self->pcmmode);
if (res < 0) return res;
/* Allocate a hwparam structure, and fill it in with configuration space */
/* Allocate a hwparam structure on the stack,
and fill it in with configuration space */
snd_pcm_hw_params_alloca(&hwparams);
res = snd_pcm_hw_params_any(self->handle, hwparams);
if (res < 0) return res;
if (res < 0)
return res;
/* Fill it in with default values. */
snd_pcm_hw_params_any(self->handle, hwparams);
@@ -110,7 +189,7 @@ static int alsapcm_setup(alsapcm_t *self) {
snd_pcm_hw_params_set_rate(self->handle, hwparams, self->rate, dir);
snd_pcm_hw_params_set_period_size(self->handle, hwparams, self->periodsize,
dir);
snd_pcm_hw_params_set_periods(self->handle,hwparams,4,0);
snd_pcm_hw_params_set_periods(self->handle, hwparams, 4, 0);
/* Write it to the device */
res = snd_pcm_hw_params(self->handle, hwparams);
@@ -118,15 +197,16 @@ static int alsapcm_setup(alsapcm_t *self) {
/* Query current settings. These may differ from the requested values,
which should therefore be sync'ed with actual values */
snd_pcm_hw_params_current(self->handle,hwparams);
snd_pcm_hw_params_current(self->handle, hwparams);
snd_pcm_hw_params_get_format(hwparams,&fmt); self->format = fmt;
snd_pcm_hw_params_get_channels(hwparams,&val); self->channels = val;
snd_pcm_hw_params_get_rate(hwparams,&val,&dir); self->rate = val;
snd_pcm_hw_params_get_period_size(hwparams,&frames,&dir);
snd_pcm_hw_params_get_format(hwparams, &fmt); self->format = fmt;
snd_pcm_hw_params_get_channels(hwparams, &val); self->channels = val;
snd_pcm_hw_params_get_rate(hwparams, &val, &dir); self->rate = val;
snd_pcm_hw_params_get_period_size(hwparams, &frames, &dir);
self->periodsize = (int) frames;
self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8;
return res;
}
@@ -134,11 +214,15 @@ static PyObject *
alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
int res;
alsapcm_t *self;
int pcmtype=0;
int pcmmode=0;
char *cardname = "default";
if (!PyArg_ParseTuple(args,"|iis",&pcmtype,&pcmmode,&cardname))
int pcmtype = SND_PCM_STREAM_PLAYBACK;
int pcmmode = 0;
char *kw[] = { "type", "mode", "card", NULL };
char *cardname = NULL;
if (!PyArg_ParseTupleAndKeywords(args, kwds, "|iiz", kw,
&pcmtype, &pcmmode, &cardname))
return NULL;
if (!(self = (alsapcm_t *)PyObject_New(alsapcm_t, &ALSAPCMType)))
return NULL;
@@ -152,17 +236,20 @@ alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
PyErr_SetString(ALSAAudioError, "Invalid PCM mode");
return NULL;
}
self->handle = 0;
self->pcmtype = pcmtype;
self->pcmmode = pcmmode;
self->cardname = strdup(cardname);
self->cardname = translate_cardname(cardname);
self->channels = 2;
self->rate = 44100;
self->format = SND_PCM_FORMAT_S16_LE;
self->periodsize = 32;
self->handle = 0;
res = alsapcm_setup(self);
res = snd_pcm_open(&(self->handle), self->cardname, self->pcmtype,
self->pcmmode);
if (res >= 0)
res = alsapcm_setup(self);
if (res < 0) {
if (self->handle) {
@@ -175,7 +262,8 @@ alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
return (PyObject *)self;
}
static void alsapcm_dealloc(alsapcm_t *self) {
static void alsapcm_dealloc(alsapcm_t *self)
{
if (self->handle) {
snd_pcm_drain(self->handle);
snd_pcm_close(self->handle);
@@ -184,6 +272,30 @@ static void alsapcm_dealloc(alsapcm_t *self) {
PyObject_Del(self);
}
static PyObject *
alsapcm_close(alsapcm_t *self, PyObject *args)
{
if (!PyArg_ParseTuple(args,":close"))
return NULL;
if (self->handle) {
Py_BEGIN_ALLOW_THREADS
snd_pcm_drain(self->handle);
snd_pcm_close(self->handle);
Py_END_ALLOW_THREADS
self->handle = 0;
}
Py_INCREF(Py_None);
return Py_None;
}
PyDoc_STRVAR(pcm_close_doc,
"close() -> None\n\
\n\
Close a PCM device.");
static PyObject *
alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
unsigned int val,val2;
@@ -194,8 +306,13 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_hw_params_current(self->handle,hwparams);
if (!PyArg_ParseTuple(args,":dumpinfo"))
return NULL;
if (!PyArg_ParseTuple(args,":dumpinfo")) return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
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)));
@@ -280,7 +397,14 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
static PyObject *
alsapcm_pcmtype(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":pcmtype")) return NULL;
if (!PyArg_ParseTuple(args,":pcmtype"))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
return PyLong_FromLong(self->pcmtype);
}
@@ -292,7 +416,14 @@ Returns either PCM_CAPTURE or PCM_PLAYBACK.");
static PyObject *
alsapcm_pcmmode(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,"pcmmode")) return NULL;
if (!PyArg_ParseTuple(args,":pcmmode"))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
return PyLong_FromLong(self->pcmmode);
}
@@ -307,7 +438,14 @@ Returns the mode of the PCM object. One of:\n\
static PyObject *
alsapcm_cardname(alsapcm_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":cardname")) return NULL;
if (!PyArg_ParseTuple(args,":cardname"))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
return PyUnicode_FromString(self->cardname);
}
@@ -321,7 +459,14 @@ static PyObject *
alsapcm_setchannels(alsapcm_t *self, PyObject *args) {
int channels;
int res;
if (!PyArg_ParseTuple(args,"i:setchannels",&channels)) return NULL;
if (!PyArg_ParseTuple(args,"i:setchannels",&channels))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
self->channels = channels;
res = alsapcm_setup(self);
if (res < 0) {
@@ -344,7 +489,14 @@ static PyObject *
alsapcm_setrate(alsapcm_t *self, PyObject *args) {
int rate;
int res;
if (!PyArg_ParseTuple(args,"i:setrate",&rate)) return NULL;
if (!PyArg_ParseTuple(args,"i:setrate",&rate))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
self->rate = rate;
res = alsapcm_setup(self);
if (res < 0) {
@@ -358,14 +510,21 @@ PyDoc_STRVAR(setrate_doc,
"setrate(rate)\n\
\n\
Set the sample rate in Hz for the device. Typical values are\n\
8000(telephony), 11025, 44100 (CD), 48000 (DVD audio) and 96000");
8000 (telephony), 11025, 44100 (CD), 48000 (DVD audio) and 96000");
static PyObject *
alsapcm_setformat(alsapcm_t *self, PyObject *args) {
int format;
int res;
if (!PyArg_ParseTuple(args,"i:setformat",&format)) return NULL;
if (!PyArg_ParseTuple(args,"i:setformat",&format))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
self->format = format;
res = alsapcm_setup(self);
if (res < 0) {
@@ -383,7 +542,14 @@ static PyObject *
alsapcm_setperiodsize(alsapcm_t *self, PyObject *args) {
int periodsize;
int res;
if (!PyArg_ParseTuple(args,"i:setperiodsize",&periodsize)) return NULL;
if (!PyArg_ParseTuple(args,"i:setperiodsize",&periodsize))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
self->periodsize = periodsize;
res = alsapcm_setup(self);
if (res < 0) {
@@ -394,12 +560,12 @@ alsapcm_setperiodsize(alsapcm_t *self, PyObject *args) {
}
PyDoc_STRVAR(setperiodsize_doc,
"setperiodsize(period)\n\
"setperiodsize(period) -> int\n\
\n\
Sets the actual period size in frames. Each write should consist of\n\
exactly this number of frames, and each read will return this number of\n\
frames (unless the device is in PCM_NONBLOCK mode, in which case it\n\
may return nothing at all)");
may return nothing at all).");
static PyObject *
alsapcm_read(alsapcm_t *self, PyObject *args) {
@@ -412,7 +578,14 @@ alsapcm_read(alsapcm_t *self, PyObject *args) {
return NULL;
}
if (!PyArg_ParseTuple(args,":read")) return NULL;
if (!PyArg_ParseTuple(args,":read"))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
if (self->pcmtype != SND_PCM_STREAM_CAPTURE) {
PyErr_SetString(ALSAAudioError,"Cannot read from playback PCM");
return NULL;
@@ -476,6 +649,11 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) {
datalen = buf.len;
#endif
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
if (datalen % self->framesize) {
PyErr_SetString(ALSAAudioError,
"Data size must be a multiple of framesize");
@@ -522,7 +700,14 @@ written at a later time.");
static PyObject *alsapcm_pause(alsapcm_t *self, PyObject *args) {
int enabled=1, res;
if (!PyArg_ParseTuple(args,"|i:pause",&enabled)) return NULL;
if (!PyArg_ParseTuple(args,"|i:pause",&enabled))
return NULL;
if (!self->handle) {
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
Py_BEGIN_ALLOW_THREADS
res = snd_pcm_pause(self->handle, enabled);
@@ -558,6 +743,7 @@ static PyMethodDef alsapcm_methods[] = {
{"read", (PyCFunction)alsapcm_read, METH_VARARGS, read_doc},
{"write", (PyCFunction)alsapcm_write, METH_VARARGS, write_doc},
{"pause", (PyCFunction)alsapcm_pause, METH_VARARGS, pause_doc},
{"close", (PyCFunction)alsapcm_close, METH_VARARGS, pcm_close_doc},
{NULL, NULL}
};
@@ -655,18 +841,25 @@ alsamixer_list(PyObject *self, PyObject *args) {
snd_mixer_selem_id_t *sid;
snd_mixer_elem_t *elem;
int err;
char *cardname = "default";
PyObject *result = PyList_New(0);
int cardidx = 0;
char cardname[32];
PyObject *result;
if (!PyArg_ParseTuple(args,"|s:mixers",&cardname)) return NULL;
if (!PyArg_ParseTuple(args,"|i:mixers",&cardidx))
return NULL;
sprintf(cardname, "hw:%d", cardidx);
snd_mixer_selem_id_alloca(&sid);
err = alsamixer_gethandle(cardname,&handle);
err = alsamixer_gethandle(cardname, &handle);
if (err < 0) {
PyErr_SetString(ALSAAudioError,snd_strerror(err));
snd_mixer_close(handle);
return NULL;
}
result = PyList_New(0);
for (elem = snd_mixer_first_elem(handle); elem;
elem = snd_mixer_elem_next(elem))
{
@@ -703,23 +896,30 @@ static PyObject *
alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
alsamixer_t *self;
int err;
char *cardname = "default";
int cardindex = 0;
char *control = "Master";
int id = 0;
snd_mixer_elem_t *elem;
int channel;
if (!PyArg_ParseTuple(args,"|sis",&control,&id,&cardname))
char *kw[] = { "control", "id", "cardindex", NULL };
if (!PyArg_ParseTupleAndKeywords(args, kwds, "|sii", kw,
&control, &id, &cardindex))
return NULL;
if (!(self = (alsamixer_t *)PyObject_New(alsamixer_t, &ALSAMixerType)))
return NULL;
err = alsamixer_gethandle(cardname,&self->handle);
self->handle = 0;
self->cardname = translate_cardidx(cardindex);
err = alsamixer_gethandle(self->cardname, &self->handle);
if (err<0) {
PyErr_SetString(ALSAAudioError,snd_strerror(err));
free(self->cardname);
return NULL;
}
self->cardname = strdup(cardname);
}
self->controlname = strdup(control);
self->controlid = id;
@@ -811,8 +1011,37 @@ static void alsamixer_dealloc(alsamixer_t *self) {
}
static PyObject *
alsamixer_cardname(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":cardname")) return NULL;
alsamixer_close(alsamixer_t *self, PyObject *args)
{
if (!PyArg_ParseTuple(args,":close"))
return NULL;
snd_mixer_close(self->handle);
free(self->cardname);
free(self->controlname);
self->handle = 0;
Py_INCREF(Py_None);
return Py_None;
}
PyDoc_STRVAR(mixer_close_doc,
"close() -> None\n\
\n\
Close a Mixer.");
static PyObject *
alsamixer_cardname(alsamixer_t *self, PyObject *args)
{
if (!PyArg_ParseTuple(args,":cardname"))
return NULL;
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "Mixer is closed");
return NULL;
}
return PyUnicode_FromString(self->cardname);
}
@@ -821,10 +1050,18 @@ PyDoc_STRVAR(mixer_cardname_doc,
\n\
Returns the name of the sound card used by this Mixer object.");
static PyObject *
alsamixer_mixer(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":mixer")) return NULL;
alsamixer_mixer(alsamixer_t *self, PyObject *args)
{
if (!PyArg_ParseTuple(args,":mixer"))
return NULL;
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "Mixer is closed");
return NULL;
}
return PyUnicode_FromString(self->controlname);
}
@@ -837,7 +1074,15 @@ for example 'Master' or 'PCM'");
static PyObject *
alsamixer_mixerid(alsamixer_t *self, PyObject *args) {
if (!PyArg_ParseTuple(args,":mixerid")) return NULL;
if (!PyArg_ParseTuple(args,":mixerid"))
return NULL;
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "Mixer is closed");
return NULL;
}
return PyLong_FromLong(self->controlid);
}
@@ -948,7 +1193,14 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args) {
char *dirstr = 0;
PyObject *result;
if (!PyArg_ParseTuple(args,"|s:getvolume",&dirstr)) return NULL;
if (!PyArg_ParseTuple(args,"|s:getvolume",&dirstr))
return NULL;
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "Mixer is closed");
return NULL;
}
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
@@ -977,7 +1229,7 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args) {
snd_mixer_selem_get_capture_volume(elem, channel, &ival);
PyList_Append(
result, PyLong_FromLong(alsamixer_getpercentage(self->cmin,
self->cmax, ival)));
self->cmax, ival)));
}
}
return result;
@@ -1001,7 +1253,14 @@ alsamixer_getrange(alsamixer_t *self, PyObject *args) {
int direction;
char *dirstr = 0;
if (!PyArg_ParseTuple(args,"|s:getrange",&dirstr)) return NULL;
if (!PyArg_ParseTuple(args,"|s:getrange",&dirstr))
return NULL;
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "Mixer is closed");
return NULL;
}
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
@@ -1059,7 +1318,14 @@ alsamixer_getenum(alsamixer_t *self, PyObject *args) {
char name[32];
PyObject *result;
if (!PyArg_ParseTuple(args, ":getenum")) return NULL;
if (!PyArg_ParseTuple(args, ":getenum"))
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)) {
@@ -1127,7 +1393,14 @@ alsamixer_getmute(alsamixer_t *self, PyObject *args) {
int i;
int ival;
PyObject *result;
if (!PyArg_ParseTuple(args,":getmute")) return NULL;
if (!PyArg_ParseTuple(args,":getmute"))
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_has_playback_switch(elem)) {
@@ -1159,7 +1432,14 @@ alsamixer_getrec(alsamixer_t *self, PyObject *args) {
int i;
int ival;
PyObject *result;
if (!PyArg_ParseTuple(args,":getrec")) return NULL;
if (!PyArg_ParseTuple(args,":getrec"))
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_has_capture_switch(elem)) {
@@ -1202,6 +1482,12 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args) {
PyErr_SetString(ALSAAudioError,"Volume must be between 0 and 100");
return NULL;
}
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "Mixer is closed");
return NULL;
}
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
@@ -1263,7 +1549,14 @@ 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;
if (!PyArg_ParseTuple(args,"i|i:setmute",&mute,&channel))
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_has_playback_switch(elem)) {
@@ -1304,7 +1597,15 @@ alsamixer_setrec(alsamixer_t *self, PyObject *args) {
int rec = 0;
int done = 0;
int channel = MIXER_CHANNEL_ALL;
if (!PyArg_ParseTuple(args,"i|i:setrec",&rec,&channel)) return NULL;
if (!PyArg_ParseTuple(args,"i|i:setrec",&rec,&channel))
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_has_capture_switch(elem)) {
@@ -1341,6 +1642,7 @@ This method will fail if the mixer has no capture switch capabilities");
static PyMethodDef alsamixer_methods[] = {
{"cardname", (PyCFunction)alsamixer_cardname, METH_VARARGS,
mixer_cardname_doc},
{"close", (PyCFunction)alsamixer_close, METH_VARARGS, mixer_close_doc},
{"mixer", (PyCFunction)alsamixer_mixer, METH_VARARGS, mixer_doc},
{"mixerid", (PyCFunction)alsamixer_mixerid, METH_VARARGS, mixerid_doc},
{"switchcap", (PyCFunction)alsamixer_switchcap, METH_VARARGS, switchcap_doc},
@@ -1415,6 +1717,7 @@ static PyTypeObject ALSAMixerType = {
/******************************************/
static PyMethodDef alsaaudio_methods[] = {
{ "cards", alsacard_list, METH_VARARGS, cards_doc},
{ "mixers", alsamixer_list, METH_VARARGS, mixers_doc},
{ 0, 0 },
};
doc/Makefile
+20 -20
View File
@@ -9,7 +9,7 @@ PAPER =
# Internal variables.
PAPEROPT_a4 = -D latex_paper_size=a4
PAPEROPT_letter = -D latex_paper_size=letter
ALLSPHINXOPTS = -d .build/doctrees $(PAPEROPT_$(PAPER)) $(SPHINXOPTS) .
ALLSPHINXOPTS = -d doctrees $(PAPEROPT_$(PAPER)) $(SPHINXOPTS) .
SFUSER = larsimmisch
@@ -27,51 +27,51 @@ help:
@echo " linkcheck to check all external links for integrity"
clean:
-rm -rf .build/*
-rm -rf html doctrees pickle htmlhelp latex changes linkcheck
html:
mkdir -p .build/html .build/doctrees
$(SPHINXBUILD) -b html $(ALLSPHINXOPTS) .build/html
mkdir -p html doctrees
$(SPHINXBUILD) -b html $(ALLSPHINXOPTS) html
@echo
@echo "Build finished. The HTML pages are in .build/html."
@echo "Build finished. The HTML pages are in html."
pickle:
mkdir -p .build/pickle .build/doctrees
$(SPHINXBUILD) -b pickle $(ALLSPHINXOPTS) .build/pickle
mkdir -p pickle doctrees
$(SPHINXBUILD) -b pickle $(ALLSPHINXOPTS) pickle
@echo
@echo "Build finished; now you can process the pickle files or run"
@echo " sphinx-web .build/pickle"
@echo " sphinx-web pickle"
@echo "to start the sphinx-web server."
web: pickle
htmlhelp:
mkdir -p .build/htmlhelp .build/doctrees
$(SPHINXBUILD) -b htmlhelp $(ALLSPHINXOPTS) .build/htmlhelp
mkdir -p htmlhelp doctrees
$(SPHINXBUILD) -b htmlhelp $(ALLSPHINXOPTS) htmlhelp
@echo
@echo "Build finished; now you can run HTML Help Workshop with the" \
".hhp project file in .build/htmlhelp."
".hhp project file in htmlhelp."
latex:
mkdir -p .build/latex .build/doctrees
$(SPHINXBUILD) -b latex $(ALLSPHINXOPTS) .build/latex
mkdir -p latex doctrees
$(SPHINXBUILD) -b latex $(ALLSPHINXOPTS) latex
@echo
@echo "Build finished; the LaTeX files are in .build/latex."
@echo "Run \`make all-pdf' or \`make all-ps' in that directory to" \
"run these through (pdf)latex."
changes:
mkdir -p .build/changes .build/doctrees
$(SPHINXBUILD) -b changes $(ALLSPHINXOPTS) .build/changes
mkdir -p changes doctrees
$(SPHINXBUILD) -b changes $(ALLSPHINXOPTS) changes
@echo
@echo "The overview file is in .build/changes."
@echo "The overview file is in changes."
linkcheck:
mkdir -p .build/linkcheck .build/doctrees
$(SPHINXBUILD) -b linkcheck $(ALLSPHINXOPTS) .build/linkcheck
mkdir -p linkcheck doctrees
$(SPHINXBUILD) -b linkcheck $(ALLSPHINXOPTS) linkcheck
@echo
@echo "Link check complete; look for any errors in the above output " \
"or in .build/linkcheck/output.txt."
"or in linkcheck/output.txt."
install:
scp -r .build/html/* $(SFUSER),pyalsaaudio@web.sourceforge.net:htdocs
scp -r ./html/* $(SFUSER),pyalsaaudio@web.sourceforge.net:htdocs
doc/conf.py
+179
View File
@@ -0,0 +1,179 @@
# -*- coding: utf-8 -*-
#
# alsaaudio documentation build configuration file, created by
# sphinx-quickstart on Sat Nov 22 00:17:09 2008.
#
# 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).
#
# All configuration values have a default value; values that are commented out
# serve to show the default value.
import sys, os
# 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
# ---------------------
# 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']
# The suffix of source filenames.
source_suffix = '.rst'
# The master toctree document.
master_doc = 'index'
# General substitutions.
project = u'alsaaudio'
copyright = u'2008, Casper Wilstrup, Lars Immisch'
# The default replacements for |version| and |release|, also used in various
# other places throughout the built documents.
#
# The short X.Y version.
version = '0.4'
# The full version, including alpha/beta/rc tags.
release = '0.4'
# 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'
# 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
# The name of the Pygments (syntax highlighting) style to use.
pygments_style = 'sphinx'
# 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'
# The name for this set of Sphinx documents. If None, it defaults to
# "<project> v<release> documentation".
#html_title = None
# 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
# 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']
# 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 = ''
# Output file base name for HTML help builder.
htmlhelp_basename = 'alsaaudiodoc'
# Options for LaTeX output
# ------------------------
# The paper size ('letter' or 'a4').
#latex_paper_size = 'letter'
# The font size ('10pt', '11pt' or '12pt').
#latex_font_size = '10pt'
# Grouping the document tree into LaTeX files. List of tuples
# (source start file, target name, title, author, document class [howto/manual]).
latex_documents = [
('index', 'alsaaudio.tex', u'alsaaudio Documentation',
u'Casper Wilstrup, 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
# 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
View File
@@ -5,6 +5,7 @@ alsaaudio documentation
:maxdepth: 2
pyalsaaudio
terminology
libalsaaudio
doc/libalsaaudio.rst
+97 -133
View File
@@ -1,84 +1,6 @@
****************************
PCM Terminology and Concepts
****************************
In order to use PCM devices it is useful to be familiar with some concepts and
terminology.
Sample
PCM audio, whether it is input or output, consists of *samples*.
A single sample represents the amplitude of one channel of sound
at a certain point in time. A lot of individual samples are
necessary to represent actual sound; for CD audio, 44100 samples
are taken every second.
Samples can be of many different sizes, ranging from 8 bit to 64
bit precision. The specific format of each sample can also vary -
they can be big endian byte integers, little endian byte integers, or
floating point numbers.
Musically, the sample size determines the dynamic range. The
dynamic range is the difference between the quietest and the
loudest signal that can be resproduced.
Frame
A frame consists of exactly one sample per channel. If there is only one
channel (Mono sound) a frame is simply a single sample. If the sound is
stereo, each frame consists of two samples, etc.
Frame size
This is the size in bytes of each frame. This can vary a lot: if each sample
is 8 bits, and we're handling mono sound, the frame size is one byte.
Similarly in 6 channel audio with 64 bit floating point samples, the frame
size is 48 bytes
Rate
PCM sound consists of a flow of sound frames. The sound rate controls how
often the current frame is replaced. For example, a rate of 8000 Hz
means that a new frame is played or captured 8000 times per second.
Data rate
This is the number of bytes, which must be recorded or provided per
second at a certain frame size and rate.
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
8000 \* 1 \* 1 = 8 kb/s or 64kbit/s. This is typically used for telephony.
At the other end of the scale, 96000 Hz, 6 channel sound with 64
bit (8 bytes) samples has a data rate of 96000 \* 6 \* 8 = 4608
kb/s (almost 5 Mb sound data per second)
Period
When the hardware processes data this is done in chunks of frames. The time
interval between each processing (A/D or D/A conversion) is known
as the period.
The size of the period has direct implication on the latency of the
sound input or output. For low-latency the period size should be
very small, while low CPU resource usage would usually demand
larger period sizes. With ALSA, the CPU utilization is not impacted
much by the period size, since the kernel layer buffers multiple
periods internally, so each period generates an interrupt and a
memory copy, but userspace can be slower and read or write multiple
periods at the same time.
Period size
This is the size of each period in Hz. *Not bytes, but Hz!.* In
:mod:`alsaaudio` the period size is set directly, and it is
therefore important to understand the significance of this
number. If the period size is configured to for example 32,
each write should contain exactly 32 frames of sound data, and each
read will return either 32 frames of data or nothing at all.
Once you understand these concepts, you will be ready to use the PCM API. Read
on.
********************
Module documentation
********************
****************
:mod:`alsaaudio`
================
****************
.. module:: alsaaudio
:platform: Linux
@@ -112,75 +34,67 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
.. % should be enclosed in \var{...}.
.. function:: mixers([cardname])
.. function:: cards()
List the available mixers. The optional *cardname* specifies which card
should be queried (this is only relevant if you have more than one sound
card). Omit to use the default sound card.
List the available cards.
.. function:: mixers([cardindex])
.. class:: PCM([type], [mode], [cardname])
List the available mixers. The optional *cardindex* specifies which card
should be queried. The default is 0.
This class is used to represent a PCM device (both playback and capture
devices). The arguments are: --- *type* - can be either PCM_CAPTURE or
PCM_PLAYBACK (default). --- *mode* - can be either PCM_NONBLOCK,
PCM_ASYNC, or PCM_NORMAL (the default). --- *cardname* - specifies
which card should be used (this is only relevant if you have more
than one sound card). Omit to use the default sound card
.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, card='default')
This class is used to represent a PCM device (both for playback and
recording - capture). The arguments are:
.. class:: Mixer([control], [id], [cardname])
* *type* - can be either ``PCM_CAPTURE`` or ``PCM_PLAYBACK`` (default).
* *mode* - can be either ``PCM_NONBLOCK``, or ``PCM_NORMAL`` (default).
* *card* - specifies the name of the card that should be used.
This class is used to access a specific ALSA mixer. The arguments are: ---
*control* - Name of the chosen mixed (default is Master). --- *id* - id of
mixer (default is 0) -- More explanation needed here --- *cardname*
specifies which card should be used (this is only relevant if you have more
than one sound card). Omit to use the default sound card.
.. class:: Mixer(control='Master', id=0, cardindex=0)
This class is used to access a specific ALSA mixer. The arguments
are:
* *control* - Name of the chosen mixed (default is 'Master').
* *id* - id of mixer -- More explanation needed here
* *cardindex* specifies which card should be used.
.. exception:: ALSAAudioError
Exception raised when an operation fails for a ALSA specific reason. The
exception argument is a string describing the reason of the failure.
.. _pcm-objects:
PCM Objects
-----------
The acronym PCM is short for Pulse Code Modulation and is the method used in
ALSA and many other places to handle playback and capture of sampled
sound data.
PCM objects in :mod:`alsaaudio` are used to do exactly that, either
play sample based sound or capture sound from some input source
(probably a microphone). The PCM object constructor takes the
PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
sound through speakers or a microphone. The PCM constructor takes the
following arguments:
.. class:: PCM(type=PCM_CAPTURE, mode=PCM_NORMAL, card='default')
.. class:: PCM([type], [mode], [cardname])
*type* - can be either ``PCM_CAPTURE`` or ``PCM_PLAYBACK`` (default).
*type* - can be either PCM_CAPTURE or PCM_PLAYBACK (default).
*mode* - can be either ``PCM_NONBLOCK``, or ``PCM_NORMAL`` (the
default). In ``PCM_NONBLOCK`` mode, calls to :func:`read` will return
immediately independent of whether there is any actual data to
read. Similarly, calls to :func:`write` will return immediately without
actually writing anything to the playout buffer if the buffer is
full [#f1]_.
*mode* - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the
default). In PCM_NONBLOCK mode, calls to read will return
immediately independent of wether there is any actual data to
read. Similarly, write calls will return immediately without
actually writing anything to the playout buffer if the buffer is full.
*card* - specifies which card should be used. This can be a string
like 'default' or a name that was returned from the :func:`cards` function.
In the current version of :mod:`alsaaudio` PCM_ASYNC is useless,
since it relies on a callback procedure, which can't be specified
through this API yet.
This will construct a PCM object with these default settings:
*cardname* - specifies which card should be used (this is only
relevant if you have more than one sound card). Omit to use the
default sound card.
This will construct a PCM object with default settings:
Sample format: PCM_FORMAT_S16_LE --- Rate: 8000 Hz --- Channels: 2 ---
Period size: 32 frames ---
* Sample format: ``PCM_FORMAT_S16_LE``
* Rate: 44100 Hz
* Channels: 2
* Period size: 32 frames
PCM objects have the following methods:
@@ -304,7 +218,7 @@ ugly clicking sounds will occur. Conversely, of too much data is
written to the device, the write function will either block
(PCM_NORMAL mode) or return zero (PCM_NONBLOCK mode).
If your program does nothing, but play sound, the easiest way is to put the
If your program does nothing but play sound, the best strategy 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.
@@ -330,7 +244,7 @@ Mixer Objects
Mixer objects provides access to the ALSA mixer API.
.. class:: Mixer([control], [id], [cardname])
.. class:: Mixer([control], [id], [cardindex])
*control* - specifies which control to manipulate using this mixer
object. The list of available controls can be found with the
@@ -339,9 +253,12 @@ Mixer objects provides access to the ALSA mixer API.
*id* - the id of the mixer control. Default is 0
*cardname* - specifies which card should be used (this is only
relevant if you have more than one sound card). Omit to use the
default sound card.
*cardindex* - specifies which card should be used[#f3]_. 0 is the
first sound card. Default is 0.
For a list of available controls, you can also use ``amixer``::
amixer -c 1
Mixer objects have the following methods:
@@ -498,7 +415,7 @@ Mixer objects have the following methods:
This method will fail if the mixer has no capture switch capabilities.
**A Note on the ALSA Mixer API**
**A rant on the ALSA Mixer API**
The ALSA mixer API is extremely complicated - and hardly documented at all.
:mod:`alsaaudio` implements a much simplified way to access this API. In
@@ -525,9 +442,56 @@ painful trial and error process.
.. _pcm-example:
ALSA Examples
-------------
Examples
--------
For now, the only examples available are the 'playbacktest.py' and the
'recordtest.py' programs included. This will change in a future version.
The following examples are provided:
* playwav.py
* recordtest.py
* playbacktest.py
All examples take the commandline option '-c <cardname>'.
To determine a valid card name, use the commandline ALSA player::
$ aplay -L
or::
$ python
>>> import alsaaudio
>>> alsaaudio.cards()
playwav.py
~~~~~~~~~~
``playwav.py`` plays a wav file. A sample wav file is
provided in the source distribution.
To test PCM playback (on your default soundcard), do::
$ python playwav.py foo.wav
recordtest.py and playbacktest.py
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
``recordtest.py`` and ``playbacktest.py`` will record and play a raw
sound file in CD quality.
To test PCM recordings (on your default soundcard), do::
$ python recordtest.py <filename>
Speak into the microphone, and interrupt the recording at any time
with ``Ctl-C``.
Play back the recording with::
$ python playbacktest.py <filename>
.. rubric:: Footnotes
.. [#f1] ALSA also allows ``PCM_ASYNC``, but this is not supported yet.
.. [#f2] But :mod:`alsaaudio` will leave any name alone that has a ':' (colon) in it.
.. [#f3] This is inconsistent with the PCM objects, which use names, but it is consistent with aplay and amixer.
doc/mixer-objects.html
-407
View File
@@ -1,407 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<html>
<head>
<link rel="STYLESHEET" href="pyalsaaudio.css" type='text/css' />
<link rel="first" href="pyalsaaudio.html" title='PyAlsaAudio' />
<link rel='contents' href='contents.html' title="Contents" />
<link rel='last' href='about.html' title='About this document...' />
<link rel='help' href='about.html' title='About this document...' />
<link rel="next" href="pcm-example.html" />
<link rel="prev" href="pcm-objects.html" />
<link rel="parent" href="module-alsaaudio.html" />
<link rel="next" href="pcm-example.html" />
<meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
<meta name='aesop' content='information' />
<title>4.3 Mixer Objects</title>
</head>
<body>
<div class="navigation">
<div id='top-navigation-panel' xml:id='top-navigation-panel'>
<table align="center" width="100%" cellpadding="0" cellspacing="2">
<tr>
<td class='online-navigation'><a rel="prev" title="4.2 pcm Objects"
href="pcm-objects.html"><img src='previous.png'
border='0' height='32' alt='Previous Page' width='32' /></a></td>
<td class='online-navigation'><a rel="parent" title="4 alsaaudio"
href="module-alsaaudio.html"><img src='up.png'
border='0' height='32' alt='Up one Level' width='32' /></a></td>
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<td align="center" width="100%">PyAlsaAudio</td>
<td class='online-navigation'><a rel="contents" title="Table of Contents"
href="contents.html"><img src='contents.png'
border='0' height='32' alt='Contents' width='32' /></a></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
<td class='online-navigation'><img src='blank.png'
border='0' height='32' alt='' width='32' /></td>
</tr></table>
<div class='online-navigation'>
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<!--End of Navigation Panel-->
<h2><a name="SECTION002430000000000000000"></a>
<a name="mixer-objects"></a>
<br>
4.3 Mixer Objects
</h2>
<p>
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-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.
<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
</dl>
<p>
Mixer objects have the following methods:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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
</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">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'
</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">mixerid</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
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-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:
<p>
<div class="center"><table class="realtable">
<thead>
<tr>
<th class="left" >Switch</th>
<th class="left" >Description</th>
</tr>
</thead>
<tbody>
<tr><td class="left" valign="baseline"><volume capabilities>'Mute'</volume></td>
<td class="left" >This mixer can be muted</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Joined Mute'</volume></td>
<td class="left" >This mixer can mute all channels at the same time</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Playback Mute'</volume></td>
<td class="left" >This mixer can mute the playback output</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Joined Playback Mute'</volume></td>
<td class="left" >Mute playback for all channels at the same time</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Capture Mute'</volume></td>
<td class="left" >Mute sound capture</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Joined Capture Mute'</volume></td>
<td class="left" >Mute sound capture for all channels at a time</td></tr>
<tr><td class="left" valign="baseline"><volume capabilities>'Capture Exclusive'</volume></td>
<td class="left" >Not quite sure what this is</td></tr></tbody>
</table></div>
<p>
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-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:
<p>
<div class="center"><table class="realtable">
<thead>
<tr>
<th class="left" >Capability</th>
<th class="left" >Description</th>
</tr>
</thead>
<tbody>
<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>
<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>
<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>
</table></div>
<p>
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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>
For enumerated controls, return the currently selected item and
the list of items available.
<p>
Returns a tuple <i>(string, list of strings)</i>.
<p>
For example, my soundcard has a Mixer called <i>Mono Output Select</i>.
Using <i>amixer</i>, I get:
<p>
<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>
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>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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.
<p>
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">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 recording.
<p>
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-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>
Returns a list with the current volume settings for each channel.
The list elements are integer percentages.
<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-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.
<p>
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
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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.
<p>
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.
</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.
<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.
<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.
<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.
<p>
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<h1><a name="SECTION002400000000000000000">
4 <tt class="module">alsaaudio</tt></a>
</h1>
<p>
<a name="module-alsaaudio"></a>
<p>
<p class="availability">Availability: <span
class="platform">Linux</span>.</p>
<p>
<p>
<p>
The <tt class="module">alsaaudio</tt> module defines functions and classes for using
ALSA.
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-2' xml:id='l2h-2' class="function">mixers</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>cardname</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
List the available mixers. The optional <var>cardname</var> specifies which
card should be queried (this is only relevant if you have more than one
sound card). Omit to use the default sound card.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><span class="typelabel">class</span>&nbsp;<tt id='l2h-4' xml:id='l2h-4' 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>
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 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>
<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.
</dd></dl>
<p>
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<h1><a name="SECTION002100000000000000000">
1 What is ALSA</a>
</h1>
<p>
The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI
functionality to the Linux operating system.
<p>
Logically ALSA consists of these components:
<ul>
<li>A set of kernel drivers.
<br>
These drivers are responsible for handling the physical sound
hardware from within the Linux kernel, and have been the standard
sound implementation in Linux since kernel version 2.5
</li>
<li>A kernel level API for manipulating the ALSA devices.
</li>
<li>A user-space C library for simplified access to the sound hardware
from userspace applications. This library is called <i>libasound</i>
and is required by all ALSA capable applications.
</li>
</ul>
<p>
More information about ALSA may be found on the project homepage
<a class="url" href="http://www.alsa-project.org">http://www.alsa-project.org</a>
<p>
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<h1><a name="SECTION002200000000000000000">
2 ALSA and Python</a>
</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).
<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.
<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 off yet.
<p>
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
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<h1><a name="SECTION002300000000000000000">
3 Installation</a>
</h1>
<p>
Note: the wrappers link with the alsasound library (from the alsa-lib
package) and need the ALSA headers for compilation. Verify that you
have /usr/lib/libasound.so and /usr/include/alsa (or
similar paths) before building.
<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:
<br><div class="verbatim"><pre>
$ python setup.py build
</pre></div>
<p>
And then as root:
<br><div class="verbatim"><pre>
# python setup.py install
</pre></div>
<p>
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<h2><a name="SECTION002410000000000000000">
4.1 PCM Terminology and Concepts</a>
</h2>
<p>
In order to use PCM devices it is useful to be familiar with some concepts and
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.
<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.
<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
<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.
<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.
<p>
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)
<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.
<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.
<p>
</dd>
</dl>
<p>
Once you understand these concepts, you will be ready to use the PCM
API. Read on.
<p>
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<h2><a name="SECTION002440000000000000000"></a><a name="pcm-example"></a>
<br>
4.4 ALSA Examples
</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.
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<h2><a name="SECTION002420000000000000000"></a>
<a name="pcm-objects"></a>
<br>
4.2 PCM Objects
</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.
<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
(probably a microphone). The PCM object constructor takes the following
arguments:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><span class="typelabel">class</span>&nbsp;<tt id='l2h-6' xml:id='l2h-6' 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>
<p>
<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.
<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 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
<p>
This will construct a PCM object with default settings:
<p>
Sample format: PCM_FORMAT_S16_LE
<br>
Rate: 8000 Hz
<br>
Channels: 2
<br>
Period size: 32 frames
<br></dl>
<p>
PCM objects have the following methods:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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
</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>format</var>)</td></tr></table></dt>
<dd>
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:
<div class="center"><table class="realtable">
<thead>
<tr>
<th class="left" >Format</th>
<th class="left" >Description</th>
</tr>
</thead>
<tbody>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_S8</formats></td>
<td class="left" >Signed 8 bit samples for each channel</td></tr>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<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>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_MPEG</formats></td>
<td class="left" >MPEG
encoded audio?</td></tr>
<tr><td class="left" valign="baseline"><formats>PCM_FORMAT_GSM</formats></td>
<td class="left" >9600 bits/s constant rate encoding for speech</td></tr></tbody>
</table></div>
<p>
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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)
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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 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.
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<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.
<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
<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.
</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.
<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).
<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.
<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.
<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.
<p>
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<p><b><font size="+2">Casper Wilstrup</font></b></p>
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+31 -4
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***************
PyAlsaAudio
***************
************
Introduction
************
:Author: Casper Wilstrup
:Author: Lars Immisch
@@ -91,7 +91,7 @@ Note: the wrappers link with the alsasound library (from the alsa-lib package)
and need the ALSA headers for compilation. Verify that you have
/usr/lib/libasound.so and /usr/include/alsa (or similar paths) before building.
On Debian (and probably Ubuntu), make sure you have libasound2-dev installed.
*On Debian (and probably Ubuntu), install libasound2-dev.*
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
@@ -105,3 +105,30 @@ To install, execute the following: --- ::
And then as root: --- ::
# python setup.py install
*******
Testing
*******
First of all, run::
$ python test.py
This is a small test suite that mostly performs consistency tests. If
it fails, please file a `bug report
<http://sourceforge.net/tracker/?group_id=120651>`_.
To test PCM recordings (on your default soundcard), verify your
microphone works, then do::
$ python recordtest.py <filename>
Speak into the microphone, and interrupt the recording at any time
with ``Ctl-C``.
Play back the recording with::
$ python playbacktest.py <filename>
doc/terminology.rst
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****************************
PCM Terminology and Concepts
****************************
In order to use PCM devices it is useful to be familiar with some concepts and
terminology.
Sample
PCM audio, whether it is input or output, consists of *samples*.
A single sample represents the amplitude of one channel of sound
at a certain point in time. A lot of individual samples are
necessary to represent actual sound; for CD audio, 44100 samples
are taken every second.
Samples can be of many different sizes, ranging from 8 bit to 64
bit precision. The specific format of each sample can also vary -
they can be big endian byte integers, little endian byte integers, or
floating point numbers.
Musically, the sample size determines the dynamic range. The
dynamic range is the difference between the quietest and the
loudest signal that can be resproduced.
Frame
A frame consists of exactly one sample per channel. If there is only one
channel (Mono sound) a frame is simply a single sample. If the sound is
stereo, each frame consists of two samples, etc.
Frame size
This is the size in bytes of each frame. This can vary a lot: if each sample
is 8 bits, and we're handling mono sound, the frame size is one byte.
Similarly in 6 channel audio with 64 bit floating point samples, the frame
size is 48 bytes
Rate
PCM sound consists of a flow of sound frames. The sound rate controls how
often the current frame is replaced. For example, a rate of 8000 Hz
means that a new frame is played or captured 8000 times per second.
Data rate
This is the number of bytes, which must be recorded or provided per
second at a certain frame size and rate.
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
8000 \* 1 \* 1 = 8 kb/s or 64kbit/s. This is typically used for telephony.
At the other end of the scale, 96000 Hz, 6 channel sound with 64
bit (8 bytes) samples has a data rate of 96000 \* 6 \* 8 = 4608
kb/s (almost 5 Mb sound data per second)
Period
When the hardware processes data this is done in chunks of frames. The time
interval between each processing (A/D or D/A conversion) is known
as the period.
The size of the period has direct implication on the latency of the
sound input or output. For low-latency the period size should be
very small, while low CPU resource usage would usually demand
larger period sizes. With ALSA, the CPU utilization is not impacted
much by the period size, since the kernel layer buffers multiple
periods internally, so each period generates an interrupt and a
memory copy, but userspace can be slower and read or write multiple
periods at the same time.
Period size
This is the size of each period in Hz. *Not bytes, but Hz!.* In
:mod:`alsaaudio` the period size is set directly, and it is
therefore important to understand the significance of this
number. If the period size is configured to for example 32,
each write should contain exactly 32 frames of sound data, and each
read will return either 32 frames of data or nothing at all.
Once you understand these concepts, you will be ready to use the PCM API. Read
on.