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26 Commits

Author SHA1 Message Date
Lars Immisch 9693a932a3 Add echo (seed for loopback.py) 2024-02-16 14:14:37 +01:00
Lars Immisch ee1c3a546b Documentation for type hints 2024-02-05 23:34:41 +00:00
Lars Immisch e4ec455ffa Add type hints 2024-02-05 23:12:44 +00:00
Oswald Buddenhagen 8fb33ddd49 improve write() underrun handling, take 2
we *really* should not paper over underruns, as they require attention.
however, the previous attempt (c2a6b6e) caused an exception to be thrown
(see #130), which was a bit excessive, and was consequently reverted
(438e52e).

so instead we make the handling consistent with what we do in read():
return the verbatim -EPIPE in this case. this can be simply ignored, and
the next write will resume the stream, so this is mostly backwards-
compatible (the failing write will be discarded and would need
repeating, but that will just cause a skip after the interruption,
which does not seem particularly relevant).

as a drive-by, again stop using snd_pcm_recover(), as it still just
obfuscates the snd_pcm_prepare() call it does in the end.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen 691c1d9b23 fix return value of PCM.write() on success (#137)
the `else` branch of the return value handling cascade got lost in
commit 438e52e, leading to us returning None on success.

rather than restoring the old code exactly, delay the construction
of the final return code object. this is more consistent with
alsapcm_read() and overall nicer.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen 061c297f4b remove stray snd_pcm_prepare() call from alsapcm_write()
this came from 438e52e, which tried to partially revert c2a6b6e, but
inserted a chunk that actually belonged to alsapcm_drop(). the latter
does not need to be restored, as we now handle SND_PCM_STATE_SETUP prior
to reading/writing.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen 8ff3e169cd unbreak read buffer overrun handling
my commit c2a6b6e broke it big time; we'd now just paper over overruns.
:}

the previous handling was fundamentally correct, needing only two
adjustments:
- to recover from drop()/drain(), we need to call snd_pcm_prepare() when
  the stream state is SND_PCM_STATE_SETUP. notably, we must not do this
  when the state is SND_PCM_STATE_XRUN.
- we should error-check the unlikely case that the recovery from an xrun
  fails.

that way we now have two snd_pcm_prepare() call sites in read(), which
looks a bit messy, but it's actually correct.

as a drive-by, simplify the return value check of snd_pcm_prepare() -
values higher than zero are impossible.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen 7d9c16618b slightly clarify docu of read() wrt. underrun 2024-02-05 23:01:30 +01:00
Oswald Buddenhagen 16345a139a make alsapcm_read()'s return value preparation clearer
... by nesting the success case into the != -EPIPE block.
2024-02-05 23:01:30 +01:00
Lars Immisch 1c730123eb pre-release updates
- update CHANGES.md
- bump version in setup.py

[Revisionist Note] This is the end of the rewritten branch. The original
history can be found in the branch main-pre-rewrite.
2024-02-02 11:42:30 +01:00
Lars Immisch 0df2e0ee6f Ignore volume events if shairplay-sync is running 2024-02-02 11:36:58 +01:00
Lars Immisch fe3fbe5376 loopback.py: bugfixes 2024-02-02 11:36:58 +01:00
Lars Immisch 42ca8acbad Add a (naive) loopback implementation (#132)
* WIP
* Open/close the playback device when idle.
  It takes a long time until it's stopped, though.
* open/close logic of playback device
* Fix opening logic, make period size divisible by 6
* Be less verbose in level info
* Extra argument for output mixer card index
  Sometimes, this cannot be deduced from the output device
* Better silence detection
* Run run_after_stop when idle on startup
2024-02-02 11:36:58 +01:00
Lars Immisch 522131123c Add PCM.polldescriptors_revents()
Will be used in the upcoming loopback implementation, but it is
worthwhile regardless.
2024-02-02 11:36:58 +01:00
Lars Immisch 43a94b3c62 Add PCM.avail()
Will be used in the upcoming loopback implementation, but it is
worthwhile regardless.
2024-02-02 11:36:58 +01:00
Lars Immisch 9637703ab5 Fix build (#133)
[Revisionist Note] This is a squashed commit formed from commits
f374adb, 3743cf5, and cd44517, still found in the main-pre-rewrite
branch. It incorporates a suggestion from PR #134.
2024-02-02 11:36:58 +01:00
Lars Immisch 438e52e3fc Restore previous behaviour of calling snd_pcm_prepare in case of XRUN (#131) 2024-02-02 11:36:58 +01:00
Lars Immisch 07ac637b1c Fix memory leaks in PCM.write() error paths on python3 2024-02-02 11:36:58 +01:00
Lars Immisch bdca4dc061 Small improvement to VolumeForwarder 2024-02-02 11:36:58 +01:00
Lars Immisch 24eef474da Refactor loopback. SCNR.
The Reactor now takes a callable, and the loopback and volume forwarder
are now implemented as callable instances, which seemed the most
Pythonic solution.
2024-02-02 11:36:58 +01:00
Lars Immisch 24d26a5161 Better error logging and comments 2024-02-02 11:36:58 +01:00
Lars Immisch f62e61f844 Add volume control forwarding
This needs the patches from (probably)
https://lkml.org/lkml/2021/3/1/419. They are already in the raspberry OS
kernel sources and the setup works on an RPi 4.
2024-02-02 11:36:58 +01:00
Lars Immisch 53f4f093e1 mixertest.py: print capture volume 2024-02-02 11:36:58 +01:00
Lars Immisch 82308f32ed Add a naive loopback implementation using select.poll()
It does work, though.
2024-02-02 11:36:58 +01:00
Lars Immisch 39d6acd3ac Handle events in alsamixer_getvolume. Closes #126
This issue can be worked around by calling mixer.handleevents() before
calling mixer.getvolume(), but it makes more sense to handle all events
before returning the volume.
2024-02-02 11:36:58 +01:00
Lars Immisch c5153db0ac Whitespace fixes
- strip trailing whitespace in several files
- fix some indentation (tabs vs. spaces)
2024-02-02 11:36:58 +01:00
10 changed files with 862 additions and 126 deletions
+5
View File
@@ -1,3 +1,8 @@
# Version 0.10.1
- revert to not throwing an exception on playback buffer underrun;
instead, return -EPIPE like `PCM.read()` does on overrun; #131
- type hints
# Version 0.10.0 # Version 0.10.0
- assorted improvements (#123 from @ossilator) - assorted improvements (#123 from @ossilator)
- support for `periods` in the `PCM` constructor. - support for `periods` in the `PCM` constructor.
+1
View File
@@ -1,4 +1,5 @@
include *.py include *.py
include alsaaudio.pyi
include CHANGES include CHANGES
include TODO include TODO
include LICENSE include LICENSE
+159 -30
View File
@@ -196,8 +196,8 @@ get_pcmtype(PyObject *obj)
static bool is_value_volume_unit(long unit) static bool is_value_volume_unit(long unit)
{ {
if (unit == VOLUME_UNITS_PERCENTAGE || if (unit == VOLUME_UNITS_PERCENTAGE ||
unit == VOLUME_UNITS_RAW || unit == VOLUME_UNITS_RAW ||
unit == VOLUME_UNITS_DB) { unit == VOLUME_UNITS_DB) {
return true; return true;
} }
return false; return false;
@@ -1364,12 +1364,23 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
buffer = PyBytes_AS_STRING(buffer_obj); buffer = PyBytes_AS_STRING(buffer_obj);
#endif #endif
// After drop() and drain(), we need to prepare the stream again.
// Note that fresh streams are already prepared by snd_pcm_hw_params().
state = snd_pcm_state(self->handle); state = snd_pcm_state(self->handle);
if ((state != SND_PCM_STATE_XRUN && state != SND_PCM_STATE_SETUP) || if ((state != SND_PCM_STATE_SETUP) ||
(res = snd_pcm_prepare(self->handle)) >= 0) { !(res = snd_pcm_prepare(self->handle))) {
Py_BEGIN_ALLOW_THREADS Py_BEGIN_ALLOW_THREADS
res = snd_pcm_readi(self->handle, buffer, self->periodsize); res = snd_pcm_readi(self->handle, buffer, self->periodsize);
Py_END_ALLOW_THREADS Py_END_ALLOW_THREADS
if (res == -EPIPE) {
// This means buffer overrun, which we need to report.
// However, we recover the stream, so the next PCM.read() will work
// again. If recovery fails (very unlikely), report that instead.
if (!(res = snd_pcm_prepare(self->handle)))
res = -EPIPE;
}
} }
if (res != -EPIPE) if (res != -EPIPE)
@@ -1385,10 +1396,10 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
Py_DECREF(buffer_obj); Py_DECREF(buffer_obj);
return NULL; return NULL;
} }
} else
{
if (res > 0 ) { sizeout = res * self->framesize;
sizeout = res * self->framesize; }
} }
if (size != sizeout) { if (size != sizeout) {
@@ -1423,11 +1434,8 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
{ {
snd_pcm_state_t state;
int res;
int datalen; int datalen;
char *data; char *data;
PyObject *rc = NULL;
#if PY_MAJOR_VERSION < 3 #if PY_MAJOR_VERSION < 3
if (!PyArg_ParseTuple(args,"s#:write", &data, &datalen)) if (!PyArg_ParseTuple(args,"s#:write", &data, &datalen))
@@ -1445,6 +1453,11 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
if (!self->handle) if (!self->handle)
{ {
PyErr_SetString(ALSAAudioError, "PCM device is closed"); PyErr_SetString(ALSAAudioError, "PCM device is closed");
#if PY_MAJOR_VERSION >= 3
PyBuffer_Release(&buf);
#endif
return NULL; return NULL;
} }
@@ -1452,34 +1465,80 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
{ {
PyErr_SetString(ALSAAudioError, PyErr_SetString(ALSAAudioError,
"Data size must be a multiple of framesize"); "Data size must be a multiple of framesize");
#if PY_MAJOR_VERSION >= 3
PyBuffer_Release(&buf);
#endif
return NULL; return NULL;
} }
state = snd_pcm_state(self->handle); int res;
if ((state != SND_PCM_STATE_XRUN && state != SND_PCM_STATE_SETUP) || // After drop() and drain(), we need to prepare the stream again.
(res = snd_pcm_prepare(self->handle)) >= 0) { // Note that fresh streams are already prepared by snd_pcm_hw_params().
snd_pcm_state_t state = snd_pcm_state(self->handle);
if ((state != SND_PCM_STATE_SETUP) ||
!(res = snd_pcm_prepare(self->handle))) {
Py_BEGIN_ALLOW_THREADS Py_BEGIN_ALLOW_THREADS
res = snd_pcm_writei(self->handle, data, datalen/self->framesize); res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
Py_END_ALLOW_THREADS Py_END_ALLOW_THREADS
if (res == -EPIPE) {
// This means buffer underrun, which we need to report.
// However, we recover the stream, so the next PCM.write() will work
// again. If recovery fails (very unlikely), report that instead.
if (!(res = snd_pcm_prepare(self->handle)))
res = -EPIPE;
}
} }
if (res == -EAGAIN) { if (res != -EPIPE)
rc = PyLong_FromLong(0);
}
else if (res < 0)
{ {
PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res), if (res == -EAGAIN)
self->cardname); {
} res = 0;
else { }
rc = PyLong_FromLong(res); else if (res < 0) {
PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
self->cardname);
#if PY_MAJOR_VERSION >= 3
PyBuffer_Release(&buf);
#endif
return NULL;
}
} }
#if PY_MAJOR_VERSION >= 3 #if PY_MAJOR_VERSION >= 3
PyBuffer_Release(&buf); PyBuffer_Release(&buf);
#endif #endif
return rc; return PyLong_FromLong(res);
}
static PyObject *
alsapcm_avail(alsapcm_t *self, PyObject *args)
{
if (!PyArg_ParseTuple(args,":avail"))
return NULL;
if (!self->handle)
{
PyErr_SetString(ALSAAudioError, "PCM device is closed");
return NULL;
}
long avail = snd_pcm_avail(self->handle);
// if (avail < 0)
// {
// PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(avail),
// self->cardname);
// return NULL;
// }
return PyLong_FromLong(avail);
} }
static PyObject *alsapcm_pause(alsapcm_t *self, PyObject *args) static PyObject *alsapcm_pause(alsapcm_t *self, PyObject *args)
@@ -1603,6 +1662,72 @@ alsapcm_polldescriptors(alsapcm_t *self, PyObject *args)
return result; return result;
} }
static PyObject *
alsapcm_polldescriptors_revents(alsapcm_t *self, PyObject *args)
{
PyObject *list_obj;
if (!PyArg_ParseTuple(args, "O!:polldescriptors_revents", &PyList_Type, &list_obj))
{
PyErr_SetString(PyExc_TypeError, "parameter must be a list.");
return NULL;
}
Py_ssize_t list_size = PyList_Size(list_obj);
struct pollfd *fds = (struct pollfd*)calloc(list_size, sizeof(struct pollfd));
if (!fds)
{
PyErr_Format(PyExc_MemoryError, "Out of memory [%s]",
self->cardname);
return NULL;
}
for (int i = 0; i < list_size; i++)
{
PyObject *tuple_obj = PyList_GetItem(list_obj, i);
if(!PyTuple_Check(tuple_obj)) {
PyErr_SetString(PyExc_TypeError, "list items must be tuples.");
free(fds);
return NULL;
}
Py_ssize_t tuple_size = PyTuple_Size(tuple_obj);
if (tuple_size != 2) {
PyErr_SetString(PyExc_TypeError, "tuples inside list must be (fd: int, mask: int)");
free(fds);
return NULL;
}
PyObject* t0 = PyTuple_GetItem(tuple_obj, 0);
PyObject* t1 = PyTuple_GetItem(tuple_obj, 1);
if (!PyLong_Check(t0) || !PyLong_Check(t1)) {
PyErr_SetString(PyExc_TypeError, "tuples inside list must be (fd: int, mask: int)");
free(fds);
return NULL;
}
// leave fds[i].event as 0 (from calloc) for now
fds[i].fd = PyLong_AS_LONG(t0);
fds[i].revents = PyLong_AS_LONG(t1);
}
unsigned short revents;
int rc = snd_pcm_poll_descriptors_revents(self->handle, fds, (unsigned short)list_size, &revents);
if (rc < 0)
{
PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(rc),
self->cardname);
free(fds);
return NULL;
}
free(fds);
return PyLong_FromLong(revents);
}
/* ALSA PCM Object Bureaucracy */ /* ALSA PCM Object Bureaucracy */
static PyMethodDef alsapcm_methods[] = { static PyMethodDef alsapcm_methods[] = {
@@ -1627,11 +1752,13 @@ static PyMethodDef alsapcm_methods[] = {
{"getchannels", (PyCFunction)alsapcm_getchannels, METH_VARARGS}, {"getchannels", (PyCFunction)alsapcm_getchannels, METH_VARARGS},
{"read", (PyCFunction)alsapcm_read, METH_VARARGS}, {"read", (PyCFunction)alsapcm_read, METH_VARARGS},
{"write", (PyCFunction)alsapcm_write, METH_VARARGS}, {"write", (PyCFunction)alsapcm_write, METH_VARARGS},
{"avail", (PyCFunction)alsapcm_avail, METH_VARARGS},
{"pause", (PyCFunction)alsapcm_pause, METH_VARARGS}, {"pause", (PyCFunction)alsapcm_pause, METH_VARARGS},
{"drop", (PyCFunction)alsapcm_drop, METH_VARARGS}, {"drop", (PyCFunction)alsapcm_drop, METH_VARARGS},
{"drain", (PyCFunction)alsapcm_drain, METH_VARARGS}, {"drain", (PyCFunction)alsapcm_drain, METH_VARARGS},
{"close", (PyCFunction)alsapcm_close, METH_VARARGS}, {"close", (PyCFunction)alsapcm_close, METH_VARARGS},
{"polldescriptors", (PyCFunction)alsapcm_polldescriptors, METH_VARARGS}, {"polldescriptors", (PyCFunction)alsapcm_polldescriptors, METH_VARARGS},
{"polldescriptors_revents", (PyCFunction)alsapcm_polldescriptors_revents, METH_VARARGS},
{NULL, NULL} {NULL, NULL}
}; };
@@ -2157,13 +2284,10 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
{ {
snd_mixer_elem_t *elem; snd_mixer_elem_t *elem;
int channel; int channel;
long ival;
PyObject *pcmtypeobj = NULL; PyObject *pcmtypeobj = NULL;
long pcmtype; long pcmtype;
int iunits = VOLUME_UNITS_PERCENTAGE; int iunits = VOLUME_UNITS_PERCENTAGE;
PyObject *result; PyObject *result = NULL;
PyObject *item;
char *kw[] = { "pcmtype", "units", NULL }; char *kw[] = { "pcmtype", "units", NULL };
if (!PyArg_ParseTupleAndKeywords(args, kwds, "|Oi:getvolume", kw, &pcmtypeobj, &iunits)) { if (!PyArg_ParseTupleAndKeywords(args, kwds, "|Oi:getvolume", kw, &pcmtypeobj, &iunits)) {
@@ -2187,6 +2311,9 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
} }
volume_units_t units = iunits; volume_units_t units = iunits;
// handle updates that may have occurred
snd_mixer_handle_events(self->handle);
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid); elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!pcmtypeobj || (pcmtypeobj == Py_None)) { if (!pcmtypeobj || (pcmtypeobj == Py_None)) {
@@ -2201,6 +2328,8 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
result = PyList_New(0); result = PyList_New(0);
for (channel = 0; channel <= SND_MIXER_SCHN_LAST; channel++) { for (channel = 0; channel <= SND_MIXER_SCHN_LAST; channel++) {
long ival;
if (pcmtype == SND_PCM_STREAM_PLAYBACK && if (pcmtype == SND_PCM_STREAM_PLAYBACK &&
snd_mixer_selem_has_playback_channel(elem, channel)) snd_mixer_selem_has_playback_channel(elem, channel))
{ {
@@ -2218,7 +2347,7 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
break; break;
} }
item = PyLong_FromLong(ival); PyObject* item = PyLong_FromLong(ival);
PyList_Append(result, item); PyList_Append(result, item);
Py_DECREF(item); Py_DECREF(item);
} }
@@ -2239,7 +2368,7 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
break; break;
} }
item = PyLong_FromLong(ival); PyObject* item = PyLong_FromLong(ival);
PyList_Append(result, item); PyList_Append(result, item);
Py_DECREF(item); Py_DECREF(item);
} }
+128
View File
@@ -0,0 +1,128 @@
from typing import list
PCM_PLAYBACK: int
PCM_CAPTURE: int
PCM_NORMAL: int
PCM_NONBLOCK: int
PCM_ASYNC: int
PCM_FORMAT_S8: int
PCM_FORMAT_U8: int
PCM_FORMAT_S16_LE: int
PCM_FORMAT_S16_BE: int
PCM_FORMAT_U16_LE: int
PCM_FORMAT_U16_BE: int
PCM_FORMAT_S24_LE: int
PCM_FORMAT_S24_BE: int
PCM_FORMAT_U24_LE: int
PCM_FORMAT_U24_BE: int
PCM_FORMAT_S32_LE: int
PCM_FORMAT_S32_BE: int
PCM_FORMAT_U32_LE: int
PCM_FORMAT_U32_BE: int
PCM_FORMAT_FLOAT_LE: int
PCM_FORMAT_FLOAT_BE: int
PCM_FORMAT_FLOAT64_LE: int
PCM_FORMAT_FLOAT64_BE: int
PCM_FORMAT_MU_LAW: int
PCM_FORMAT_A_LAW: int
PCM_FORMAT_IMA_ADPCM: int
PCM_FORMAT_MPEG: int
PCM_FORMAT_GSM: int
PCM_FORMAT_S24_3LE: int
PCM_FORMAT_S24_3BE: int
PCM_FORMAT_U24_3LE: int
PCM_FORMAT_U24_3BE: int
PCM_TSTAMP_NONE: int
PCM_TSTAMP_ENABLE: int
PCM_TSTAMP_TYPE_GETTIMEOFDAY: int
PCM_TSTAMP_TYPE_MONOTONIC: int
PCM_TSTAMP_TYPE_MONOTONIC_RAW: int
PCM_FORMAT_DSD_U8: int
PCM_FORMAT_DSD_U16_LE: int
PCM_FORMAT_DSD_U32_LE: int
PCM_FORMAT_DSD_U32_BE: int
PCM_STATE_OPEN: int
PCM_STATE_SETUP: int
PCM_STATE_PREPARED: int
PCM_STATE_RUNNING: int
PCM_STATE_XRUN: int
PCM_STATE_DRAINING: int
PCM_STATE_PAUSED: int
PCM_STATE_SUSPENDED: int
PCM_STATE_DISCONNECTED: int
MIXER_CHANNEL_ALL: int
MIXER_SCHN_UNKNOWN: int
MIXER_SCHN_FRONT_LEFT: int
MIXER_SCHN_FRONT_RIGHT: int
MIXER_SCHN_REAR_LEFT: int
MIXER_SCHN_REAR_RIGHT: int
MIXER_SCHN_FRONT_CENTER: int
MIXER_SCHN_WOOFER: int
MIXER_SCHN_SIDE_LEFT: int
MIXER_SCHN_SIDE_RIGHT: int
MIXER_SCHN_REAR_CENTER: int
MIXER_SCHN_MONO: int
VOLUME_UNITS_PERCENTAGE: int
VOLUME_UNITS_RAW: int
VOLUME_UNITS_DB: int
def pcms(pcmtype:int) -> list[str]: ...
def cards() -> list[str]: ...
def mixers(cardindex:int=-1, device:str='default') -> list[str]: ...
def asoundlib_version() -> str: ...
class PCM:
def __init__(type:int=PCM_PLAYBACK, mode:int=PCM_NORMAL, rate:int=44100, channels:int=2,
format:int=PCM_FORMAT_S16_LE, periodsize:int=32, periods:int=4,
device:str='default', cardindex:int=-1) -> PCM: ...
def info() -> dict: ...
def pcmtype() -> int: ...
def pcmmode() -> int: ...
def cardname() -> str: ...
def setchannels(nchannels: int) -> None: ...
def setrate(rate: int) -> None: ...
def setformat(format: int) -> int: ...
def setperiodsize(period: int) -> int: ...
def dumpinfo() -> None: ...
def state() -> int: ...
def read() -> tuple[int, bytes]: ...
def write(data:bytes) -> int: ...
def pause(enable:bool=True) -> int: ...
def drop() -> int: ...
def drain() -> int: ...
def polldescriptors() -> list[tuple[int, int]]: ...
def set_tstamp_mode(mode:int=PCM_TSTAMP_ENABLE) -> None: ...
def get_tstamp_mode() -> int: ...
def set_tstamp_type(type:int=PCM_TSTAMP_TYPE_GETTIMEOFDAY) -> None: ...
def get_tstamp_type() -> int: ...
def htimestamp() -> tuple[int, int, int]: ...
class Mixer:
def __init__(control:str='Master', id:int=0, cardindex:int=-1, device:str='default') -> Mixer: ...
def cardname() -> str: ...
def mixer() -> str: ...
def mixerid() -> int: ...
def switchcap() -> int: ...
def volumecap() -> int: ...
def getenum() -> tuple[ str, list[str]]: ...
def setenum(index:int) -> None: ...
def getrange(pcmtype:int=PCM_PLAYBACK, units:int=VOLUME_UNITS_RAW) -> tuple[int, int]: ...
def getvolume(pcmtype:int=PCM_PLAYBACK, units:int=VOLUME_UNITS_PERCENTAGE) -> int: ...
def setvolume(volume:int, pcmtype:int=PCM_PLAYBACK, units:int=VOLUME_UNITS_PERCENTAGE, channel:int|None=None) -> None: ...
def getmute() -> list[int]: ...
def setmute(mute:bool, channel:int|None=None) -> None: ...
def getrec() -> list[int]: ...
def setrec(capture:int, channel:int|None=None) -> None: ...
def polldescriptors() -> list[tuple[int, int]]: ...
def close() -> None: ...
+89 -85
View File
@@ -10,14 +10,14 @@
The :mod:`alsaaudio` module defines functions and classes for using ALSA. The :mod:`alsaaudio` module defines functions and classes for using ALSA.
.. function:: pcms(pcmtype=PCM_PLAYBACK) .. function:: pcms(pcmtype:int=PCM_PLAYBACK) ->list[str]
List available PCM devices by name. List available PCM devices by name.
Arguments are: Arguments are:
* *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK` * *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default). (default).
**Note:** **Note:**
@@ -34,7 +34,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
*New in 0.8* *New in 0.8*
.. function:: cards() .. function:: cards() -> list[str]
List the available ALSA cards by name. This function is only moderately List the available ALSA cards by name. This function is only moderately
useful. If you want to see a list of available PCM devices, use :func:`pcms` useful. If you want to see a list of available PCM devices, use :func:`pcms`
@@ -96,18 +96,18 @@ PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
sound through speakers or a microphone. The PCM constructor takes the sound through speakers or a microphone. The PCM constructor takes the
following arguments: following arguments:
.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1) .. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1) -> PCM
This class is used to represent a PCM device (either for playback and This class is used to represent a PCM device (either for playback and
recording). The arguments are: recording). The arguments are:
* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK` * *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default). (default).
* *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL` * *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
(default). (default).
* *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``. * *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``.
* *channels* - the number of channels. The default value is 2 (stereo). * *channels* - the number of channels. The default value is 2 (stereo).
* *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures. * *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures.
The default value is :const:`PCM_FORMAT_S16_LE`. The default value is :const:`PCM_FORMAT_S16_LE`.
========================= =============== ========================= ===============
@@ -179,24 +179,24 @@ following arguments:
PCM objects have the following methods: PCM objects have the following methods:
.. method:: PCM.info() .. method:: PCM.info() -> dict
The info function returns a dictionary containing the configuration of a PCM device. As ALSA takes into account limitations of the hardware and software devices the configuration achieved might not correspond to the values used during creation. There is therefore a need to check the realised configuration before processing the sound coming from the device or before sending sound to a device. A small subset of parameters can be set, but cannot be queried. These parameters are stored by alsaaudio and returned as they were given by the user, to distinguish them from parameters retrieved from ALSA these parameters have a name prefixed with **" (call value) "**. Yet another set of properties derives directly from the hardware and can be obtained through ALSA. The info function returns a dictionary containing the configuration of a PCM device. As ALSA takes into account limitations of the hardware and software devices the configuration achieved might not correspond to the values used during creation. There is therefore a need to check the realised configuration before processing the sound coming from the device or before sending sound to a device. A small subset of parameters can be set, but cannot be queried. These parameters are stored by alsaaudio and returned as they were given by the user, to distinguish them from parameters retrieved from ALSA these parameters have a name prefixed with **" (call value) "**. Yet another set of properties derives directly from the hardware and can be obtained through ALSA.
=========================== ============================= ================================================================== =========================== ============================= ==================================================================
Key Description (Reference) Type Key Description (Reference) Type
=========================== ============================= ================================================================== =========================== ============================= ==================================================================
name PCM():device string name PCM():device string
card_no *index of card* integer (negative indicates device not associable with a card) card_no *index of card* integer (negative indicates device not associable with a card)
device_no *index of PCM device* integer device_no *index of PCM device* integer
subdevice_no *index of PCM subdevice* integer subdevice_no *index of PCM subdevice* integer
state *name of PCM state* string state *name of PCM state* string
access_type *name of PCM access type* string access_type *name of PCM access type* string
(call value) type PCM():type integer (call value) type PCM():type integer
(call value) type_name PCM():type string (call value) type_name PCM():type string
(call value) mode PCM():mode integer (call value) mode PCM():mode integer
(call value) mode_name PCM():mode string (call value) mode_name PCM():mode string
format PCM():format integer format PCM():format integer
format_name PCM():format string format_name PCM():format string
format_description PCM():format string format_description PCM():format string
subformat_name *name of PCM subformat* string subformat_name *name of PCM subformat* string
@@ -219,13 +219,13 @@ PCM objects have the following methods:
can_mmap_sample_resolution *hw: sample-resol. mmap* boolean (True: hardware supported) can_mmap_sample_resolution *hw: sample-resol. mmap* boolean (True: hardware supported)
can_pause *hw: pause* boolean (True: hardware supported) can_pause *hw: pause* boolean (True: hardware supported)
can_resume *hw: resume* boolean (True: hardware supported) can_resume *hw: resume* boolean (True: hardware supported)
can_sync_start *hw: synchronized start* boolean (True: hardware supported) can_sync_start *hw: synchronized start* boolean (True: hardware supported)
=========================== ============================= ================================================================== =========================== ============================= ==================================================================
The italicized descriptions give a summary of the "full" description as it can be found in the `ALSA documentation <https://www.alsa-project.org/alsa-doc>`_. "hw:": indicates that the property indicated relates to the hardware. Parameters passed to the PCM object during instantation are prefixed with "PCM():", they are described there for the keyword argument indicated after "PCM():". The italicized descriptions give a summary of the "full" description as it can be found in the `ALSA documentation <https://www.alsa-project.org/alsa-doc>`_. "hw:": indicates that the property indicated relates to the hardware. Parameters passed to the PCM object during instantation are prefixed with "PCM():", they are described there for the keyword argument indicated after "PCM():".
.. method:: PCM.pcmtype() .. method:: PCM.pcmtype() -> int
Returns the type of PCM object. Either :const:`PCM_CAPTURE` or Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
:const:`PCM_PLAYBACK`. :const:`PCM_PLAYBACK`.
@@ -267,37 +267,29 @@ PCM objects have the following methods:
Returns a dictionary of supported format codes (integers) keyed by Returns a dictionary of supported format codes (integers) keyed by
their standard ALSA names (strings). their standard ALSA names (strings).
.. method:: PCM.setchannels(nchannels) .. method:: PCM.setchannels(nchannels: int) -> int
.. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`. .. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
.. method:: PCM.setrate(rate) .. method:: PCM.setrate(rate: int) -> int
.. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`. .. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
.. method:: PCM.setformat(format) .. method:: PCM.setformat(format: int) -> int
.. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`. .. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
.. method:: PCM.setperiodsize(period) .. method:: PCM.setperiodsize(period: int) -> int
.. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`. .. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
.. method:: PCM.info()
Returns a dictionary with the PCM object's configured parameters.
Values are retrieved from the ALSA library if they are available;
otherwise they represent those stored by pyalsaaudio, and their keys
are prefixed with ' (call value) '.
*New in 0.9.1* *New in 0.9.1*
.. method:: PCM.dumpinfo() .. method:: PCM.dumpinfo() -> None
Dumps the PCM object's configured parameters to stdout. Dumps the PCM object's configured parameters to stdout.
.. method:: PCM.state() .. method:: PCM.state() -> int
Returs the current state of the stream, which can be one of Returs the current state of the stream, which can be one of
:const:`PCM_STATE_OPEN` (this should not actually happen), :const:`PCM_STATE_OPEN` (this should not actually happen),
@@ -312,54 +304,66 @@ PCM objects have the following methods:
*New in 0.10* *New in 0.10*
.. method:: PCM.read() .. method:: PCM.read() -> tuple[int, bytes]
In :const:`PCM_NORMAL` mode, this function blocks until a full period is In :const:`PCM_NORMAL` mode, this function blocks until a full period is
available, and then returns a tuple (length,data) where *length* is available, and then returns a tuple (length,data) where *length* is
the number of frames of captured data, and *data* is the captured the number of frames of captured data, and *data* is the captured
sound frames as a string. The length of the returned data will be sound frames as a string. The length of the returned data will be
periodsize\*framesize bytes. periodsize\*framesize bytes.
In :const:`PCM_NONBLOCK` mode, the call will not block, but will return In :const:`PCM_NONBLOCK` mode, the call will not block, but will return
``(0,'')`` if no new period has become available since the last ``(0,'')`` if no new period has become available since the last
call to read. call to read.
In case of an overrun, this function will return a negative size: :const:`-EPIPE`. In case of a buffer overrun, this function will return the negative
This indicates that data was lost, even if the operation itself succeeded. size :const:`-EPIPE`, and no data is read.
Try using a larger periodsize. This indicates that data was lost. To resume capturing, just call read
again, but note that the stream was already corrupted.
To avoid the problem in the future, try using a larger period size
and/or more periods, at the cost of higher latency.
.. method:: PCM.write(data) .. method:: PCM.write(data: bytes) -> int
Writes (plays) the sound in data. The length of data *must* be a Writes (plays) the sound in data. The length of data *must* be a
multiple of the frame size, and *should* be exactly the size of a multiple of the frame size, and *should* be exactly the size of a
period. If less than 'period size' frames are provided, the actual period. If less than 'period size' frames are provided, the actual
playout will not happen until more data is written. playout will not happen until more data is written.
If the device is not in :const:`PCM_NONBLOCK` mode, this call will block if If the data was successfully written, the call returns the size of the
the kernel buffer is full, and until enough sound has been played data *in frames*.
to allow the sound data to be buffered. The call always returns the
size of the data provided. If the device is not in :const:`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.
In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a
return value of zero, if the buffer is full. In this case, the data return value of zero, if the buffer is full. In this case, the data
should be written at a later time. should be written again at a later time.
In case of a buffer underrun, this function will return the negative
size :const:`-EPIPE`, and no data is written.
At this point, the playback was already corrupted. If you want to play
the data nonetheless, call write again with the same data.
To avoid the problem in the future, try using a larger period size
and/or more periods, at the cost of higher latency.
Note that this call completing means only that the samples were buffered Note that this call completing means only that the samples were buffered
in the kernel, and playout will continue afterwards. Make sure that the in the kernel, and playout will continue afterwards. Make sure that the
stream is drained before discarding the PCM handle. stream is drained before discarding the PCM handle.
.. method:: PCM.pause([enable=True]) .. method:: PCM.pause([enable=True]) -> int
If *enable* is :const:`True`, playback or capture is paused. If *enable* is :const:`True`, playback or capture is paused.
Otherwise, playback/capture is resumed. Otherwise, playback/capture is resumed.
.. method:: PCM.drop() .. method:: PCM.drop() -> int
Stop the stream and drop residual buffered frames. Stop the stream and drop residual buffered frames.
*New in 0.9* *New in 0.9*
.. method:: PCM.drain() .. method:: PCM.drain() -> int
For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
and then stop the stream. In :const:`PCM_NORMAL` mode, and then stop the stream. In :const:`PCM_NORMAL` mode,
@@ -369,44 +373,37 @@ PCM objects have the following methods:
*New in 0.10* *New in 0.10*
.. method:: PCM.close() .. method:: PCM.polldescriptors() -> list[tuple[int, int]]
Closes the PCM device.
For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
this function blocks until all pending playback is drained.
.. method:: PCM.polldescriptors()
Returns a list of tuples of *(file descriptor, eventmask)* that can be Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the PCM with *select.poll*. used to wait for changes on the PCM with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module. :const:`select` module.
.. method:: PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE]) .. method:: PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE]) -> None
Set the ALSA timestamp mode on the device. The mode argument can be set to Set the ALSA timestamp mode on the device. The mode argument can be set to
either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`. either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
.. method:: PCM.get_tstamp_mode() .. method:: PCM.get_tstamp_mode() -> int
Return the integer value corresponding to the ALSA timestamp mode. The Return the integer value corresponding to the ALSA timestamp mode. The
return value can be either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`. return value can be either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
.. method:: PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY]) .. method:: PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY]) -> None
Set the ALSA timestamp mode on the device. The type argument Set the ALSA timestamp mode on the device. The type argument
can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`, can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`. :const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
.. method:: PCM.get_tstamp_type() .. method:: PCM.get_tstamp_type() -> int
Return the integer value corresponding to the ALSA timestamp type. The Return the integer value corresponding to the ALSA timestamp type. The
return value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`, return value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`. :const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
.. method:: PCM.htimestamp() .. method:: PCM.htimestamp() -> tuple[int, int, int]
Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*. Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*.
@@ -433,9 +430,16 @@ PCM objects have the following methods:
update. update.
================================= =========================================== ================================= ===========================================
.. method:: PCM.close() -> None
Closes the PCM device.
For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
this function blocks until all pending playback is drained.
**A few hints on using PCM devices for playback** **A few hints on using PCM devices for playback**
The most common reason for problems with playback of PCM audio is that writes The most common reason for problems with playback of PCM audio is that writes
to PCM devices must *exactly* match the data rate of the device. to PCM devices must *exactly* match the data rate of the device.
If too little data is written to the device, it will underrun, and If too little data is written to the device, it will underrun, and
@@ -468,12 +472,12 @@ Mixer Objects
Mixer objects provides access to the ALSA mixer API. Mixer objects provides access to the ALSA mixer API.
.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default') .. class:: Mixer(control='Master', id=0, cardindex=-1, device='default') -> Mixer
Arguments are: Arguments are:
* *control* - specifies which control to manipulate using this mixer * *control* - specifies which control to manipulate using this mixer
object. The list of available controls can be found with the object. The list of available controls can be found with the
:mod:`alsaaudio`.\ :func:`mixers` function. The default value is :mod:`alsaaudio`.\ :func:`mixers` function. The default value is
``'Master'`` - other common controls may be ``'Master Mono'``, ``'PCM'``, ``'Master'`` - other common controls may be ``'Master Mono'``, ``'PCM'``,
``'Line'``, etc. ``'Line'``, etc.
@@ -483,7 +487,7 @@ Mixer objects provides access to the ALSA mixer API.
* *cardindex* - specifies which card should be used. If this argument * *cardindex* - specifies which card should be used. If this argument
is given, the device name is constructed like this: 'hw:*cardindex*' and is given, the device name is constructed like this: 'hw:*cardindex*' and
the `device` keyword argument is ignored. ``0`` is the the `device` keyword argument is ignored. ``0`` is the
first sound card. first sound card.
* *device* - the name of the device on which the mixer resides. The default * *device* - the name of the device on which the mixer resides. The default
value is ``'default'``. value is ``'default'``.
@@ -495,20 +499,20 @@ Mixer objects provides access to the ALSA mixer API.
Mixer objects have the following methods: Mixer objects have the following methods:
.. method:: Mixer.cardname() .. method:: Mixer.cardname() -> str
Return the name of the sound card used by this Mixer object Return the name of the sound card used by this Mixer object
.. method:: Mixer.mixer() .. method:: Mixer.mixer() -> str
Return the name of the specific mixer controlled by this object, For example Return the name of the specific mixer controlled by this object, For example
``'Master'`` or ``'PCM'`` ``'Master'`` or ``'PCM'``
.. method:: Mixer.mixerid() .. method:: Mixer.mixerid() -> int
Return the ID of the ALSA mixer controlled by this object. Return the ID of the ALSA mixer controlled by this object.
.. method:: Mixer.switchcap() .. method:: Mixer.switchcap() -> int
Returns a list of the switches which are defined by this specific mixer. Returns a list of the switches which are defined by this specific mixer.
Possible values in this list are: Possible values in this list are:
@@ -520,7 +524,7 @@ Mixer objects have the following methods:
'Joined Mute' This mixer can mute all channels at the same time 'Joined Mute' This mixer can mute all channels at the same time
'Playback Mute' This mixer can mute the playback output 'Playback Mute' This mixer can mute the playback output
'Joined Playback Mute' Mute playback for all channels at the same time} 'Joined Playback Mute' Mute playback for all channels at the same time}
'Capture Mute' Mute sound capture 'Capture Mute' Mute sound capture
'Joined Capture Mute' Mute sound capture for all channels at a time} 'Joined Capture Mute' Mute sound capture for all channels at a time}
'Capture Exclusive' Not quite sure what this is 'Capture Exclusive' Not quite sure what this is
====================== ================ ====================== ================
@@ -528,7 +532,7 @@ Mixer objects have the following methods:
To manipulate these switches use the :meth:`setrec` or To manipulate these switches use the :meth:`setrec` or
:meth:`setmute` methods :meth:`setmute` methods
.. method:: Mixer.volumecap() .. method:: Mixer.volumecap() -> int
Returns a list of the volume control capabilities of this Returns a list of the volume control capabilities of this
mixer. Possible values in the list are: mixer. Possible values in the list are:
@@ -544,7 +548,7 @@ Mixer objects have the following methods:
'Joined Capture Volume' Manipulate sound capture volume for all channels at a time 'Joined Capture Volume' Manipulate sound capture volume for all channels at a time
======================== ================ ======================== ================
.. method:: Mixer.getenum() .. method:: Mixer.getenum() -> tuple[ str, list[str]]
For enumerated controls, return the currently selected item and the list of For enumerated controls, return the currently selected item and the list of
items available. items available.
@@ -570,13 +574,13 @@ Mixer objects have the following methods:
This method will return an empty tuple if the mixer is not an enumerated This method will return an empty tuple if the mixer is not an enumerated
control. control.
.. method:: Mixer.setenum(index) .. method:: Mixer.setenum(index:int) -> None
For enumerated controls, sets the currently selected item. For enumerated controls, sets the currently selected item.
*index* is an index into the list of available enumerated items returned *index* is an index into the list of available enumerated items returned
by :func:`getenum`. by :func:`getenum`.
.. method:: Mixer.getrange(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_RAW) .. method:: Mixer.getrange(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_RAW) -> tuple[int, int]
Return the volume range of the ALSA mixer controlled by this object. Return the volume range of the ALSA mixer controlled by this object.
The value is a tuple of integers whose meaning is determined by the The value is a tuple of integers whose meaning is determined by the
@@ -590,7 +594,7 @@ Mixer objects have the following methods:
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`, The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`. :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.getvolume(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE) .. method:: Mixer.getvolume(pcmtype:int=PCM_PLAYBACK, units:int=VOLUME_UNITS_PERCENTAGE) -> int
Returns a list with the current volume settings for each channel. The list Returns a list with the current volume settings for each channel. The list
elements are integers whose meaning is determined by the *units* argument. elements are integers whose meaning is determined by the *units* argument.
@@ -603,7 +607,7 @@ Mixer objects have the following methods:
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`, The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`. :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.setvolume(volume, channel=None, pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE) .. method:: Mixer.setvolume(volume:int, pcmtype:int=PCM_PLAYBACK, units:int=VOLUME_UNITS_PERCENTAGE, channel:int|None) -> None
Change the current volume settings for this mixer. The *volume* argument Change the current volume settings for this mixer. The *volume* argument
is an integer whose meaning is determined by the *units* argument. is an integer whose meaning is determined by the *units* argument.
@@ -620,14 +624,14 @@ Mixer objects have the following methods:
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`, The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`. :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.getmute() .. method:: Mixer.getmute() -> list[int]
Return a list indicating the current mute setting for each channel. Return a list indicating the current mute setting for each channel.
0 means not muted, 1 means muted. 0 means not muted, 1 means muted.
This method will fail if the mixer has no playback switch capabilities. This method will fail if the mixer has no playback switch capabilities.
.. method:: Mixer.setmute(mute, [channel]) .. method:: Mixer.setmute(mute:bool, channel:int|None=None) -> None
Sets the mute flag to a new value. The *mute* argument is either 0 for not Sets the mute flag to a new value. The *mute* argument is either 0 for not
muted, or 1 for muted. muted, or 1 for muted.
@@ -637,14 +641,14 @@ Mixer objects have the following methods:
This method will fail if the mixer has no playback mute capabilities This method will fail if the mixer has no playback mute capabilities
.. method:: Mixer.getrec() .. method:: Mixer.getrec() -> list[int]
Return a list indicating the current record mute setting for each channel. Return a list indicating the current record mute setting for each channel.
0 means not recording, 1 means recording. 0 means not recording, 1 means recording.
This method will fail if the mixer has no capture switch capabilities. This method will fail if the mixer has no capture switch capabilities.
.. method:: Mixer.setrec(capture, [channel]) .. method:: Mixer.setrec(capture:int, channel:int|None=None) -> None
Sets the capture mute flag to a new value. The *capture* argument Sets the capture mute flag to a new value. The *capture* argument
is either 0 for no capture, or 1 for capture. is either 0 for no capture, or 1 for capture.
@@ -654,21 +658,21 @@ Mixer objects have the following methods:
This method will fail if the mixer has no capture switch capabilities. This method will fail if the mixer has no capture switch capabilities.
.. method:: Mixer.polldescriptors() .. method:: Mixer.polldescriptors() -> list[tuple[int, int]]
Returns a list of tuples of *(file descriptor, eventmask)* that can be Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the mixer with *select.poll*. used to wait for changes on the mixer with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module. :const:`select` module.
.. method:: Mixer.handleevents() .. method:: Mixer.handleevents() -> int
Acknowledge events on the :func:`polldescriptors` file descriptors Acknowledge events on the :func:`polldescriptors` file descriptors
to prevent subsequent polls from returning the same events again. to prevent subsequent polls from returning the same events again.
Returns the number of events that were acknowledged. Returns the number of events that were acknowledged.
.. method:: Mixer.close() .. method:: Mixer.close() -> None
Closes the Mixer device. Closes the Mixer device.
@@ -707,7 +711,7 @@ The following example are provided:
* `playbacktest.py` * `playbacktest.py`
* `mixertest.py` * `mixertest.py`
All examples (except `mixertest.py`) accept the commandline option All examples (except `mixertest.py`) accept the commandline option
*-c <cardname>*. *-c <cardname>*.
To determine a valid card name, use the commandline ALSA player:: To determine a valid card name, use the commandline ALSA player::
@@ -722,12 +726,12 @@ or::
>>> alsaaudio.pcms() >>> alsaaudio.pcms()
mixertest.py accepts the commandline options *-d <device>* and mixertest.py accepts the commandline options *-d <device>* and
*-c <cardindex>*. *-c <cardindex>*.
playwav.py playwav.py
~~~~~~~~~~ ~~~~~~~~~~
**playwav.py** plays a wav file. **playwav.py** plays a wav file.
To test PCM playback (on your default soundcard), run:: To test PCM playback (on your default soundcard), run::
@@ -775,7 +779,7 @@ The output might look like this::
'Mix' 'Mix'
'Mix Mono' 'Mix Mono'
With a single argument - the *control*, it will display the settings of With a single argument - the *control*, it will display the settings of
that control; for example:: that control; for example::
$ ./mixertest.py Master $ ./mixertest.py Master
@@ -784,7 +788,7 @@ that control; for example::
Channel 0 volume: 61% Channel 0 volume: 61%
Channel 1 volume: 61% Channel 1 volume: 61%
With two arguments, the *control* and a *parameter*, it will set the With two arguments, the *control* and a *parameter*, it will set the
parameter on the mixer:: parameter on the mixer::
$ ./mixertest.py Master mute $ ./mixertest.py Master mute
+40
View File
@@ -0,0 +1,40 @@
#!/usr/bin/env python3
# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
import alsaaudio
import select
def echo(inpcm, outpcm):
q = list()
# setup the synchronous event loop
# See https://docs.python.org/3/library/select.html#poll-objects for background
reactor = select.poll()
infd, inmask = inpcm.polldecriptors()
outfd, outmask = outpcm.polldescriptors()
write_started = False
def write():
data = q.pop()
written = outpcm.write(data)
if written < len(data):
q.insert(0, data[written:])
reactor.register(infd, inmask)
reactor.register(outfd, outmask)
while True:
events = reactor.poll()
for fd, event in events:
if event == select.POLLIN and fd == infd:
data = inpcm.read()
q.append(data)
if not write_started:
write()
write_started = True
elif event == select.POLLOUT and fd == outfd:
if not q:
return
write()
+397
View File
@@ -0,0 +1,397 @@
#!/usr/bin/env python3
# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
import sys
import select
import logging
import re
import struct
import subprocess
from datetime import datetime, timedelta
from alsaaudio import (PCM, pcms, PCM_PLAYBACK, PCM_CAPTURE, PCM_NONBLOCK, Mixer,
PCM_STATE_OPEN, PCM_STATE_SETUP, PCM_STATE_PREPARED, PCM_STATE_RUNNING, PCM_STATE_XRUN, PCM_STATE_DRAINING,
PCM_STATE_PAUSED, PCM_STATE_SUSPENDED, ALSAAudioError)
from argparse import ArgumentParser
poll_names = {
select.POLLIN: 'POLLIN',
select.POLLPRI: 'POLLPRI',
select.POLLOUT: 'POLLOUT',
select.POLLERR: 'POLLERR',
select.POLLHUP: 'POLLHUP',
select.POLLRDHUP: 'POLLRDHUP',
select.POLLNVAL: 'POLLNVAL'
}
state_names = {
PCM_STATE_OPEN: 'PCM_STATE_OPEN',
PCM_STATE_SETUP: 'PCM_STATE_SETUP',
PCM_STATE_PREPARED: 'PCM_STATE_PREPARED',
PCM_STATE_RUNNING: 'PCM_STATE_RUNNING',
PCM_STATE_XRUN: 'PCM_STATE_XRUN',
PCM_STATE_DRAINING: 'PCM_STATE_DRAINING',
PCM_STATE_PAUSED: 'PCM_STATE_PAUSED',
PCM_STATE_SUSPENDED: 'PCM_STATE_SUSPENDED'
}
def poll_desc(mask):
return '|'.join([poll_names[bit] for bit, name in poll_names.items() if mask & bit])
class PollDescriptor(object):
'''File Descriptor, event mask and a name for logging'''
def __init__(self, name, fd, mask):
self.name = name
self.fd = fd
self.mask = mask
def as_tuple(self):
return (self.fd, self.mask)
@classmethod
def from_alsa_object(cls, name, alsaobject, mask=None):
# TODO maybe refactor: we ignore objects that have more then one polldescriptor
fd, alsamask = alsaobject.polldescriptors()[0]
if mask is None:
mask = alsamask
return cls(name, fd, mask)
class Loopback(object):
'''Loopback state and event handling'''
def __init__(self, capture, playback_args, volume_handler, run_after_stop=None, run_before_start=None):
self.playback_args = playback_args
self.playback = None
self.volume_handler = volume_handler
self.capture_started = None
self.last_capture_event = None
self.capture = capture
self.capture_pd = PollDescriptor.from_alsa_object('capture', capture)
self.run_after_stop = run_after_stop.split(' ')
self.run_before_start = run_before_start.split(' ')
self.run_after_stop_did_run = False
self.waitBeforeOpen = False
self.queue = []
self.period_size = 0
self.silent_periods = 0
@staticmethod
def compute_energy(data):
values = struct.unpack(f'{len(data)//2}h', data)
e = 0
for v in values:
e = e + v * v
return e
@staticmethod
def run_command(cmd):
if cmd:
rc = subprocess.run(cmd)
if rc.returncode:
logging.warning(f'run {cmd}, return code {rc.returncode}')
else:
logging.info(f'run {cmd}, return code {rc.returncode}')
def register(self, reactor):
reactor.register_timeout_handler(self.timeout_handler)
reactor.register(self.capture_pd, self)
def start(self):
# start reading data
size, data = self.capture.read()
if size:
self.queue.append(data)
def timeout_handler(self):
if self.playback and self.capture_started:
if self.last_capture_event:
if datetime.now() - self.last_capture_event > timedelta(seconds=2):
logging.info('timeout - closing playback device')
self.playback.close()
self.playback = None
self.capture_started = None
if self.volume_handler:
self.volume_handler.stop()
self.run_command(self.run_after_stop)
return
self.waitBeforeOpen = False
if not self.run_after_stop_did_run and not self.playback:
if self.volume_handler:
self.volume_handler.stop()
self.run_command(self.run_after_stop)
self.run_after_stop_did_run = True
def pop(self):
if len(self.queue):
return self.queue.pop()
else:
return None
def handle_capture_event(self, eventmask, name):
'''called when data is available for reading'''
self.last_capture_event = datetime.now()
size, data = self.capture.read()
if not size:
logging.warning(f'capture event but no data')
return False
energy = self.compute_energy(data)
logging.debug(f'energy: {energy}')
# the usecase is a USB capture device where we get perfect silence when it's idle
if energy == 0:
self.silent_periods = self.silent_periods + 1
# turn off playback after two seconds of silence
# 2 channels * 2 seconds * 2 bytes per sample
fps = self.playback_args['rate'] * 8 // (self.playback_args['periodsize'] * self.playback_args['periods'])
logging.debug(f'{self.silent_periods} of {fps} silent periods: {self.playback}')
if self.silent_periods > fps and self.playback:
logging.info(f'closing playback due to silence')
self.playback.close()
self.playback = None
if self.volume_handler:
self.volume_handler.stop()
self.run_command(self.run_after_stop)
self.run_after_stop_did_run = True
if not self.playback:
return
else:
self.silent_periods = 0
if not self.playback:
if self.waitBeforeOpen:
return False
try:
if self.volume_handler:
self.volume_handler.start()
self.run_command(self.run_before_start)
self.playback = PCM(**self.playback_args)
self.period_size = self.playback.info()['period_size']
logging.info(f'opened playback device with period_size {self.period_size}')
except ALSAAudioError as e:
logging.info('opening PCM playback device failed: %s', e)
self.waitBeforeOpen = True
return False
self.capture_started = datetime.now()
logging.info(f'{self.playback} capture started: {self.capture_started}')
self.queue.append(data)
if len(self.queue) <= 2:
logging.info(f'buffering: {len(self.queue)}')
return False
try:
data = self.pop()
if data:
space = self.playback.avail()
written = self.playback.write(data)
logging.debug(f'wrote {written} bytes while space was {space}')
except ALSAAudioError:
logging.error('underrun', exc_info=1)
return True
def __call__(self, fd, eventmask, name):
if fd == self.capture_pd.fd:
real_mask = self.capture.polldescriptors_revents([self.capture_pd.as_tuple()])
if real_mask:
return self.handle_capture_event(real_mask, name)
else:
logging.debug('null capture event')
return False
else:
real_mask = self.playback.polldescriptors_revents([self.playback_pd.as_tuple()])
if real_mask:
return self.handle_playback_event(real_mask, name)
else:
logging.debug('null playback event')
return False
class VolumeForwarder(object):
'''Volume control event handling'''
def __init__(self, capture_control, playback_control):
self.playback_control = playback_control
self.capture_control = capture_control
self.active = True
def start(self):
self.active = True
if self.volume:
self.volume = playback_control.setvolume(self.volume)
def stop(self):
self.active = False
self.volume = self.playback_control.getvolume(pcmtype=PCM_CAPTURE)[0]
def __call__(self, fd, eventmask, name):
if not self.active:
return
volume = self.capture_control.getvolume(pcmtype=PCM_CAPTURE)
# indicate that we've handled the event
self.capture_control.handleevents()
logging.info(f'{name} adjusting volume to {volume}')
if volume:
self.playback_control.setvolume(volume[0])
class Reactor(object):
'''A wrapper around select.poll'''
def __init__(self):
self.poll = select.poll()
self.descriptors = {}
self.timeout_handlers = set()
def register(self, polldescriptor, callable):
logging.debug(f'registered {polldescriptor.name}: {poll_desc(polldescriptor.mask)}')
self.descriptors[polldescriptor.fd] = (polldescriptor, callable)
self.poll.register(polldescriptor.fd, polldescriptor.mask)
def unregister(self, polldescriptor):
self.poll.unregister(polldescriptor.fd)
del self.descriptors[polldescriptor.fd]
def register_timeout_handler(self, callable):
self.timeout_handlers.add(callable)
def unregister_timeout_handler(self, callable):
self.timeout_handlers.remove(callable)
def run(self):
last_timeout_ev = datetime.now()
while True:
# poll for a bit, then send a timeout to registered handlers
events = self.poll.poll(0.25)
for fd, ev in events:
polldescriptor, handler = self.descriptors[fd]
# very chatty - log all events
# logging.debug(f'{polldescriptor.name}: {poll_desc(ev)} ({ev})')
handler(fd, ev, polldescriptor.name)
if datetime.now() - last_timeout_ev > timedelta(seconds=0.25):
for t in self.timeout_handlers:
t()
last_timeout_ev = datetime.now()
if __name__ == '__main__':
logging.basicConfig(format='%(asctime)s %(levelname)s %(message)s', level=logging.INFO)
parser = ArgumentParser(description='ALSA loopback (with volume forwarding)')
playback_pcms = pcms(pcmtype=PCM_PLAYBACK)
capture_pcms = pcms(pcmtype=PCM_CAPTURE)
if not playback_pcms:
logging.error('no playback PCM found')
sys.exit(2)
if not capture_pcms:
logging.error('no capture PCM found')
sys.exit(2)
parser.add_argument('-d', '--debug', action='store_true')
parser.add_argument('-i', '--input', default=capture_pcms[0])
parser.add_argument('-o', '--output', default=playback_pcms[0])
parser.add_argument('-r', '--rate', type=int, default=44100)
parser.add_argument('-c', '--channels', type=int, default=2)
parser.add_argument('-p', '--periodsize', type=int, default=444) # must be divisible by 6 for 44k1
parser.add_argument('-P', '--periods', type=int, default=2)
parser.add_argument('-I', '--input-mixer', help='Control of the input mixer, can contain the card index, e.g. Digital:2')
parser.add_argument('-O', '--output-mixer', help='Control of the output mixer, can contain the card index, e.g. PCM:1')
parser.add_argument('-A', '--run-after-stop', help='command to run when the capture device is idle/silent')
parser.add_argument('-B', '--run-before-start', help='command to run when the capture device becomes active')
parser.add_argument('-V', '--volume', help='Initial volume (default is leave unchanged)')
args = parser.parse_args()
if args.debug:
logging.getLogger().setLevel(logging.DEBUG)
playback_args = {
'type': PCM_PLAYBACK,
'mode': PCM_NONBLOCK,
'device': args.output,
'rate': args.rate,
'channels': args.channels,
'periodsize': args.periodsize,
'periods': args.periods
}
reactor = Reactor()
# If args.input_mixer and args.output_mixer are set, forward the capture volume to the playback volume.
# The usecase is a capture device that is implemented using g_audio, i.e. the Linux USB gadget driver.
# When a USB device (eg. an iPad) is connected to this machine, its volume events will go to the volume control
# of the output device
capture = None
playback = None
volume_handler = None
if args.input_mixer and args.output_mixer:
re_mixer = re.compile(r'([a-zA-Z0-9]+):?([0-9+])?')
input_mixer_card = None
m = re_mixer.match(args.input_mixer)
if m:
input_mixer = m.group(1)
if m.group(2):
input_mixer_card = int(m.group(2))
else:
parser.print_usage()
sys.exit(1)
output_mixer_card = None
m = re_mixer.match(args.output_mixer)
if m:
output_mixer = m.group(1)
if m.group(2):
output_mixer_card = int(m.group(2))
else:
parser.print_usage()
sys.exit(1)
if input_mixer_card is None:
capture = PCM(type=PCM_CAPTURE, mode=PCM_NONBLOCK, device=args.input, rate=args.rate,
channels=args.channels, periodsize=args.periodsize, periods=args.periods)
input_mixer_card = capture.info()['card_no']
if output_mixer_card is None:
playback = PCM(**playback_args)
output_mixer_card = playback.info()['card_no']
playback.close()
playback_control = Mixer(control=output_mixer, cardindex=int(output_mixer_card))
capture_control = Mixer(control=input_mixer, cardindex=int(input_mixer_card))
volume_handler = VolumeForwarder(capture_control, playback_control)
reactor.register(PollDescriptor.from_alsa_object('capture_control', capture_control, select.POLLIN), volume_handler)
if args.volume and playback_control:
playback_control.setvolume(int(args.volume))
loopback = Loopback(capture, playback_args, volume_handler, args.run_after_stop, args.run_before_start)
loopback.register(reactor)
loopback.start()
reactor.run()
+8 -3
View File
@@ -72,8 +72,13 @@ def show_mixer(name, kwargs):
volumes = mixer.getvolume() volumes = mixer.getvolume()
volumes_dB = mixer.getvolume(units=alsaaudio.VOLUME_UNITS_DB) volumes_dB = mixer.getvolume(units=alsaaudio.VOLUME_UNITS_DB)
for i in range(len(volumes)): for i in range(len(volumes)):
print("Channel %i volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0)) print("Channel %i playback volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
volumes = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE)
volumes_dB = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE, units=alsaaudio.VOLUME_UNITS_DB)
for i in range(len(volumes)):
print("Channel %i capture volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
try: try:
mutes = mixer.getmute() mutes = mixer.getmute()
for i in range(len(mutes)): for i in range(len(mutes)):
@@ -113,7 +118,7 @@ def set_mixer(name, args, kwargs):
mixer.setmute(1, channel) mixer.setmute(1, channel)
else: else:
mixer.setmute(0, channel) mixer.setmute(0, channel)
elif args in ['rec','unrec']: elif args in ['rec','unrec']:
# Enable/disable recording # Enable/disable recording
if args == 'rec': if args == 'rec':
+3 -3
View File
@@ -8,7 +8,7 @@ from setuptools import setup
from setuptools.extension import Extension from setuptools.extension import Extension
from sys import version from sys import version
pyalsa_version = '0.10.0' pyalsa_version = '0.10.1'
if __name__ == '__main__': if __name__ == '__main__':
setup( setup(
@@ -29,12 +29,12 @@ if __name__ == '__main__':
'License :: OSI Approved :: Python Software Foundation License', 'License :: OSI Approved :: Python Software Foundation License',
'Operating System :: POSIX :: Linux', 'Operating System :: POSIX :: Linux',
'Programming Language :: Python :: 2', 'Programming Language :: Python :: 2',
'Programming Language :: Python :: 3', 'Programming Language :: Python :: 3',
'Topic :: Multimedia :: Sound/Audio', 'Topic :: Multimedia :: Sound/Audio',
'Topic :: Multimedia :: Sound/Audio :: Mixers', 'Topic :: Multimedia :: Sound/Audio :: Mixers',
'Topic :: Multimedia :: Sound/Audio :: Players', 'Topic :: Multimedia :: Sound/Audio :: Players',
'Topic :: Multimedia :: Sound/Audio :: Capture/Recording', 'Topic :: Multimedia :: Sound/Audio :: Capture/Recording',
], ],
ext_modules=[Extension('alsaaudio',['alsaaudio.c'], ext_modules=[Extension('alsaaudio',['alsaaudio.c'],
libraries=['asound'])] libraries=['asound'])]
) )
+32 -5
View File
@@ -11,6 +11,7 @@
import unittest import unittest
import alsaaudio import alsaaudio
import warnings import warnings
from contextlib import closing
# we can't test read and write well - these are tested otherwise # we can't test read and write well - these are tested otherwise
PCMMethods = [ PCMMethods = [
@@ -20,7 +21,7 @@ PCMMethods = [
] ]
PCMDeprecatedMethods = [ PCMDeprecatedMethods = [
('setchannels', (2,)), ('setchannels', (2,)),
('setrate', (44100,)), ('setrate', (44100,)),
('setformat', (alsaaudio.PCM_FORMAT_S8,)), ('setformat', (alsaaudio.PCM_FORMAT_S8,)),
('setperiodsize', (320,)) ('setperiodsize', (320,))
@@ -49,10 +50,10 @@ class MixerTest(unittest.TestCase):
def testMixer(self): def testMixer(self):
"""Open the default Mixers and the Mixers on every card""" """Open the default Mixers and the Mixers on every card"""
for c in alsaaudio.card_indexes(): for c in alsaaudio.card_indexes():
mixers = alsaaudio.mixers(cardindex=c) mixers = alsaaudio.mixers(cardindex=c)
for m in mixers: for m in mixers:
mixer = alsaaudio.Mixer(m, cardindex=c) mixer = alsaaudio.Mixer(m, cardindex=c)
mixer.close() mixer.close()
@@ -73,7 +74,7 @@ class MixerTest(unittest.TestCase):
mixer.close() mixer.close()
def testMixerClose(self): def testMixerClose(self):
"""Run common Mixer methods on a closed object and verify it raises an """Run common Mixer methods on a closed object and verify it raises an
error""" error"""
mixers = alsaaudio.mixers() mixers = alsaaudio.mixers()
@@ -133,7 +134,7 @@ class PCMTest(unittest.TestCase):
pcm = alsaaudio.PCM(card='default') pcm = alsaaudio.PCM(card='default')
except alsaaudio.ALSAAudioError: except alsaaudio.ALSAAudioError:
pass pass
# Verify we got a DepreciationWarning # Verify we got a DepreciationWarning
self.assertEqual(len(w), 1, "PCM(card='default') expected a warning" ) self.assertEqual(len(w), 1, "PCM(card='default') expected a warning" )
self.assertTrue(issubclass(w[-1].category, DeprecationWarning), "PCM(card='default') expected a DeprecationWarning") self.assertTrue(issubclass(w[-1].category, DeprecationWarning), "PCM(card='default') expected a DeprecationWarning")
@@ -156,5 +157,31 @@ class PCMTest(unittest.TestCase):
self.assertEqual(len(w), 1, method + " expected a warning") self.assertEqual(len(w), 1, method + " expected a warning")
self.assertTrue(issubclass(w[-1].category, DeprecationWarning), method + " expected a DeprecationWarning") self.assertTrue(issubclass(w[-1].category, DeprecationWarning), method + " expected a DeprecationWarning")
class PollDescriptorArgsTest(unittest.TestCase):
'''Test invalid args for polldescriptors_revents (takes a list of tuples of 2 integers)'''
def testArgsNoList(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents('foo')
def testArgsListButNoTuples(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents(['foo', 1])
def testArgsListButInvalidTuples(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents([('foo', 'bar')])
def testArgsListTupleWrongLength(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents([(1, )])
with self.assertRaises(TypeError):
pcm.polldescriptors_revents([(1, 2, 3)])
if __name__ == '__main__': if __name__ == '__main__':
unittest.main() unittest.main()