Experimenting with autoapi

getchannels/setchannels should be together IMO

CHANGES.md

@@ -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
 - assorted improvements (#123 from @ossilator)
   - support for `periods` in the `PCM` constructor.

MANIFEST.in

@@ -1,5 +1,6 @@
 include *.py
+include *.pyi
 include CHANGES
 include TODO
 include LICENSE
-recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json  Makefile
+recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json  Makefile

README.md

@@ -2,7 +2,7 @@
 
 For documentation, see http://larsimmisch.github.io/pyalsaaudio/
 
-> Author: Casper Wilstrup (cwi@aves.dk)  
+> Author: Casper Wilstrup (cwi@aves.dk)
 > Maintainer: Lars Immisch (lars@ibp.de)
 
 This package contains wrappers for accessing the
@@ -45,12 +45,12 @@ First, get the sources and change to the source directory:
   $ git clone https://github.com/larsimmisch/pyalsaaudio.git
   $ cd pyalsaaudio
 ```
-  
+
 Then, build:
 ```
   $ python setup.py build
 ```
-  
+
 And install:
 ```
   $ sudo python setup.py install

alsaaudio.c

@@ -196,13 +196,19 @@ get_pcmtype(PyObject *obj)
 static bool is_value_volume_unit(long unit)
 {
 	if (unit == VOLUME_UNITS_PERCENTAGE ||
-	    unit == VOLUME_UNITS_RAW ||
+		unit == VOLUME_UNITS_RAW ||
-	    unit == VOLUME_UNITS_DB) {
+		unit == VOLUME_UNITS_DB) {
 		return true;
 	}
 	return false;
 }
 
+const char * const alsacard_list_doc = R"(.. function:: cards() -> list[str]
+
+   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`
+   instead.)";
+
 static PyObject *
 alsacard_list(PyObject *self, PyObject *args)
 {
@@ -403,7 +409,7 @@ static int alsapcm_setup(alsapcm_t *self)
 	snd_pcm_hw_params_get_period_size(hwparams, &self->periodsize, &dir);
 	snd_pcm_hw_params_get_periods(hwparams, &self->periods, &dir);
 
-	self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8;
+	self->framesize = self->channels * snd_pcm_format_physical_width(self->format)/8;
 
 	return res;
 }
@@ -598,6 +604,12 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args)
 	val = snd_pcm_hw_params_get_sbits(hwparams);
 	printf("significant bits = %d\n", val);
 
+	val = snd_pcm_format_width(self->format);
+	printf("nominal bits = %d\n", val);
+
+	val = snd_pcm_format_physical_width(self->format);
+	printf("physical bits = %d\n", val);
+
 	val = snd_pcm_hw_params_is_batch(hwparams);
 	printf("is batch = %d\n", val);
 
@@ -778,6 +790,16 @@ alsapcm_info(alsapcm_t *self, PyObject *args)
 	PyDict_SetItemString(info,"significant_bits", value);
 	Py_DECREF(value);
 
+	val = snd_pcm_format_width(self->format);
+	value=PyLong_FromUnsignedLong((unsigned long) val);
+	PyDict_SetItemString(info,"nominal_bits", value);
+	Py_DECREF(value);
+
+	val = snd_pcm_format_physical_width(self->format);
+	value=PyLong_FromUnsignedLong((unsigned long) val);
+	PyDict_SetItemString(info,"physical_bits", value);
+	Py_DECREF(value);
+
 	val = snd_pcm_hw_params_is_batch(hwparams);
 	value=PyBool_FromLong((unsigned long) val);
 	PyDict_SetItemString(info,"is_batch", value);
@@ -1154,6 +1176,37 @@ alsapcm_getchannels(alsapcm_t *self,PyObject *args)
 	return out;
 }
 
+static PyObject *
+alsapcm_setchannels(alsapcm_t *self, PyObject *args)
+{
+	int channels, saved;
+	int res;
+
+	if (!PyArg_ParseTuple(args,"i:setchannels", &channels))
+		return NULL;
+
+	if (!self->handle) {
+		PyErr_SetString(ALSAAudioError, "PCM device is closed");
+		return NULL;
+	}
+
+	PyErr_WarnEx(PyExc_DeprecationWarning,
+				 "This function is deprecated. "
+				 "Please use the named parameter `channels` to `PCM()` instead", 1);
+
+	saved = self->channels;
+	self->channels = channels;
+	res = alsapcm_setup(self);
+	if (res < 0)
+	{
+		self->channels = saved;
+		PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
+					 self->cardname);
+		return NULL;
+	}
+	return PyLong_FromLong(self->channels);
+}
+
 static PyObject *
 alsapcm_pcmtype(alsapcm_t *self, PyObject *args)
 {
@@ -1196,37 +1249,6 @@ alsapcm_cardname(alsapcm_t *self, PyObject *args)
 	return PyUnicode_FromString(self->cardname);
 }
 
-static PyObject *
-alsapcm_setchannels(alsapcm_t *self, PyObject *args)
-{
-	int channels, saved;
-	int res;
-
-	if (!PyArg_ParseTuple(args,"i:setchannels", &channels))
-		return NULL;
-
-	if (!self->handle) {
-		PyErr_SetString(ALSAAudioError, "PCM device is closed");
-		return NULL;
-	}
-
-	PyErr_WarnEx(PyExc_DeprecationWarning,
-				 "This function is deprecated. "
-				 "Please use the named parameter `channels` to `PCM()` instead", 1);
-
-	saved = self->channels;
-	self->channels = channels;
-	res = alsapcm_setup(self);
-	if (res < 0)
-	{
-		self->channels = saved;
-		PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
-					 self->cardname);
-		return NULL;
-	}
-	return PyLong_FromLong(self->channels);
-}
-
 static PyObject *
 alsapcm_setrate(alsapcm_t *self, PyObject *args)
 {
@@ -1364,12 +1386,23 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
 	buffer = PyBytes_AS_STRING(buffer_obj);
 #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);
-	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
 		res = snd_pcm_readi(self->handle, buffer, self->periodsize);
 		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)
@@ -1385,10 +1418,10 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
 			Py_DECREF(buffer_obj);
 			return NULL;
 		}
-	}
+		else
-
+		{
-	if (res > 0 ) {
+			sizeout = res * self->framesize;
-		sizeout = res * self->framesize;
+		}
 	}
 
 	if (size != sizeout) {
@@ -1423,11 +1456,8 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
 
 static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
 {
-	snd_pcm_state_t state;
-	int res;
 	int datalen;
 	char *data;
-	PyObject *rc = NULL;
 
 #if PY_MAJOR_VERSION < 3
 	if (!PyArg_ParseTuple(args,"s#:write", &data, &datalen))
@@ -1445,6 +1475,11 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
 	if (!self->handle)
 	{
 		PyErr_SetString(ALSAAudioError, "PCM device is closed");
+
+#if PY_MAJOR_VERSION >= 3
+		PyBuffer_Release(&buf);
+#endif
+
 		return NULL;
 	}
 
@@ -1452,34 +1487,80 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
 	{
 		PyErr_SetString(ALSAAudioError,
 						"Data size must be a multiple of framesize");
+
+#if PY_MAJOR_VERSION >= 3
+		PyBuffer_Release(&buf);
+#endif
 		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
 		res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
 		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
 	PyBuffer_Release(&buf);
 #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)
@@ -1603,12 +1684,79 @@ alsapcm_polldescriptors(alsapcm_t *self, PyObject *args)
 	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 */
 
 static PyMethodDef alsapcm_methods[] = {
 	{"pcmtype", (PyCFunction)alsapcm_pcmtype, METH_VARARGS},
 	{"pcmmode", (PyCFunction)alsapcm_pcmmode, METH_VARARGS},
 	{"cardname", (PyCFunction)alsapcm_cardname, METH_VARARGS},
+	{"getchannels", (PyCFunction)alsapcm_getchannels, METH_VARARGS},
 	{"setchannels", (PyCFunction)alsapcm_setchannels, METH_VARARGS},
 	{"setrate", (PyCFunction)alsapcm_setrate, METH_VARARGS},
 	{"setformat", (PyCFunction)alsapcm_setformat, METH_VARARGS},
@@ -1624,14 +1772,15 @@ static PyMethodDef alsapcm_methods[] = {
 	{"getformats", (PyCFunction)alsapcm_getformats, METH_VARARGS},
 	{"getratebounds", (PyCFunction)alsapcm_getratemaxmin, METH_VARARGS},
 	{"getrates", (PyCFunction)alsapcm_getrates, METH_VARARGS},
-	{"getchannels", (PyCFunction)alsapcm_getchannels, METH_VARARGS},
 	{"read", (PyCFunction)alsapcm_read, METH_VARARGS},
 	{"write", (PyCFunction)alsapcm_write, METH_VARARGS},
+	{"avail", (PyCFunction)alsapcm_avail, METH_VARARGS},
 	{"pause", (PyCFunction)alsapcm_pause, METH_VARARGS},
 	{"drop", (PyCFunction)alsapcm_drop, METH_VARARGS},
 	{"drain", (PyCFunction)alsapcm_drain, METH_VARARGS},
 	{"close", (PyCFunction)alsapcm_close, METH_VARARGS},
 	{"polldescriptors", (PyCFunction)alsapcm_polldescriptors, METH_VARARGS},
+	{"polldescriptors_revents", (PyCFunction)alsapcm_polldescriptors_revents, METH_VARARGS},
 	{NULL, NULL}
 };
 
@@ -2157,13 +2306,10 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
 {
 	snd_mixer_elem_t *elem;
 	int channel;
-	long ival;
 	PyObject *pcmtypeobj = NULL;
 	long pcmtype;
 	int iunits = VOLUME_UNITS_PERCENTAGE;
-	PyObject *result;
+	PyObject *result = NULL;
-	PyObject *item;
-
 	char *kw[] = { "pcmtype", "units", NULL };
 
 	if (!PyArg_ParseTupleAndKeywords(args, kwds, "|Oi:getvolume", kw, &pcmtypeobj, &iunits)) {
@@ -2187,6 +2333,9 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
 	}
 	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);
 
 	if (!pcmtypeobj || (pcmtypeobj == Py_None)) {
@@ -2201,6 +2350,8 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
 	result = PyList_New(0);
 
 	for (channel = 0; channel <= SND_MIXER_SCHN_LAST; channel++) {
+		long ival;
+
 		if (pcmtype == SND_PCM_STREAM_PLAYBACK &&
 			snd_mixer_selem_has_playback_channel(elem, channel))
 		{
@@ -2218,7 +2369,7 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
 				break;
 			}
 
-			item = PyLong_FromLong(ival);
+			PyObject* item = PyLong_FromLong(ival);
 			PyList_Append(result, item);
 			Py_DECREF(item);
 		}
@@ -2239,7 +2390,7 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args, PyObject *kwds)
 				break;
 			}
 
-			item = PyLong_FromLong(ival);
+			PyObject* item = PyLong_FromLong(ival);
 			PyList_Append(result, item);
 			Py_DECREF(item);
 		}
@@ -2934,7 +3085,7 @@ static PyTypeObject ALSAMixerType = {
 static PyMethodDef alsaaudio_methods[] = {
 	{ "card_indexes", (PyCFunction)alsacard_list_indexes, METH_VARARGS},
 	{ "card_name", (PyCFunction)alsacard_name, METH_VARARGS},
-	{ "cards", (PyCFunction)alsacard_list, METH_VARARGS},
+	{ "cards", (PyCFunction)alsacard_list, METH_VARARGS, alsacard_list_doc },
 	{ "pcms", (PyCFunction)alsapcm_list, METH_VARARGS|METH_KEYWORDS},
 	{ "mixers", (PyCFunction)alsamixer_list, METH_VARARGS|METH_KEYWORDS},
 	{ 0, 0 },

alsaaudio.pyi

@@ -0,0 +1,135 @@
+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 close() -> None: ...
+	def dumpinfo() -> None: ...
+	def info() -> dict: ...
+	def state() -> int: ...
+	def htimestamp() -> tuple[int, 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 getformats() -> dict: ...
+	def getratebounds() -> tuple[int, int]: ...
+	def getrates() -> int | tuple[int, int] | list[int]: ...
+	def getchannels() -> list[int]: ...
+	def setchannels(nchannels: int) -> None: ...
+	def pcmtype() -> int: ...
+	def pcmmode() -> int: ...
+	def cardname() -> str: ...
+	def setrate(rate: int) -> None: ...
+	def setformat(format: int) -> int: ...
+	def setperiodsize(period: int) -> int: ...
+	def read() -> tuple[int, bytes]: ...
+	def write(data: bytes) -> int: ...
+	def avail() -> int: ...
+	def pause(enable: bool = True) -> int: ...
+	def drop() -> int: ...
+	def drain() -> int: ...
+	def polldescriptors() -> list[tuple[int, int]]: ...
+	def polldescriptors_revents(descriptors: list[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 close() -> None: ...
+	def mixer() -> str: ...
+	def mixerid() -> int: ...
+	def switchcap() -> int: ...
+	def volumecap() -> int: ...
+	def getvolume(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int: ...
+	def getrange(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]: ...
+	def getenum() -> tuple[str, list[str]]: ...
+	def getmute() -> list[int]: ...
+	def getrec() -> list[int]: ...
+	def setvolume(volume: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None: ...
+	def setenum(index: int) -> None: ...
+	def setmute(mute: bool, channel: (int | None) = None) -> None: ...
+	def setrec(capture: int, channel: (int | None) = None) -> None: ...
+	def polldescriptors() -> list[tuple[int, int]]: ...
+	def handleevents() -> int: ...
+

doc/README.md

@@ -26,7 +26,7 @@ Don't forget to update the documentation.
 The documentation is published through the `gh-pages` branch.
 
 To publish the documentation, you need to clone the `gh-pages` branch of this repository into
-`doc/gh-pages`. In `doc`, do: 
+`doc/gh-pages`. In `doc`, do:
 
     git clone -b gh-pages git@github.com:larsimmisch/pyalsaaudio.git gh-pages
 

doc/conf.py

@@ -34,7 +34,9 @@ from setup import pyalsa_version
 # Add any Sphinx extension module names here, as strings. They can be
 # extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
 # ones.
-extensions = []
+extensions = ['autoapi.extension']
+
+autoapi_dirs = ['..']
 
 # Add any paths that contain templates here, relative to this directory.
 templates_path = ['_templates']
@@ -67,7 +69,7 @@ release = version
 #
 # This is also used if you do content translation via gettext catalogs.
 # Usually you set "language" from the command line for these cases.
-language = None
+language = 'en'
 
 # List of patterns, relative to source directory, that match files and
 # directories to ignore when looking for source files.

doc/libalsaaudio.rst

@@ -10,14 +10,14 @@
 
 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.
 
    Arguments are:
 
    * *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).  
+     (default).
 
    **Note:**
 
@@ -34,19 +34,13 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
 
    *New in 0.8*
 
-.. function:: cards()
-
-   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`
-   instead.
-
 ..
    Omitted by intention due to being superseded by cards():
 
    .. function:: card_indexes()
    .. function:: card_name()
 
-.. function:: mixers(cardindex=-1, device='default')
+.. function:: mixers(cardindex: int = -1, device: str = 'default') -> list[str]
 
    List the available mixers. The arguments are:
 
@@ -82,7 +76,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
      changed. Since 0.8, this functions returns the mixers for the default
      device, not the mixers for the first card.
 
-.. function:: asoundlib_version()
+.. function:: asoundlib_version() -> str
 
    Return a Python string containing the ALSA version found.
 
@@ -96,18 +90,20 @@ 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_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1)
+.. class:: PCM(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
 
-   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 or
-   recording). The arguments are:
+   recording). The constructor's arguments are:
 
    * *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).  
+     (default).
    * *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``.
    * *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`.
 
    =========================  ===============
@@ -157,7 +153,15 @@ following arguments:
 
      **Note:** This should not be used, as it bypasses most of ALSA's configuration.
 
-   This will construct a PCM object with the given settings.
+   The defaults mentioned above are values passed by :mod:alsaaudio
+   to ALSA, not anything internal to ALSA.
+
+   **Note:** For default and non-default values alike, there is no
+   guarantee that a PCM device supports the requested configuration,
+   and ALSA may pick realizable values which it believes to be closest
+   to the request. Therefore, after creating a PCM object, it is
+   necessary to verify whether its realized configuration is acceptable.
+   The :func:info method can be used to query it.
 
    *Changed in 0.10:*
 
@@ -179,63 +183,89 @@ following arguments:
 
 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.
+   Returns a dictionary containing the configuration of a PCM device.
-   
-   ===========================  =============================  ==================================================================
-        Key                      Description (Reference)            Type       
-   ===========================  =============================  ==================================================================
-   name                         PCM():device                      string
-   card_no                      *index of card*                   integer  (negative indicates device not associable with a card)
-   device_no                    *index of PCM device*             integer 
-   subdevice_no                 *index of PCM subdevice*          integer 
-   state                        *name of PCM state*               string
-   access_type                  *name of PCM access type*         string
-   (call value) type            PCM():type                        integer
-   (call value) type_name       PCM():type                        string
-   (call value) mode            PCM():mode                        integer
-   (call value) mode_name       PCM():mode                        string
-   format                       PCM():format                      integer 
-   format_name                  PCM():format                      string
-   format_description           PCM():format                      string
-   subformat_name               *name of PCM subformat*           string
-   subformat_description        *description of subformat*        string
-   channels                     PCM():channels                    integer
-   rate                         PCM():rate                        integer (Hz)
-   period_time                  *period duration*                 integer (:math:`\mu s`)
-   period_size                  PCM():period_size                 integer (frames)
-   buffer_time                  *buffer time*                     integer (:math:`\mu s`) (negative indicates error)
-   buffer_size                  *buffer size*                     integer (frames) (negative indicates error)
-   get_periods                  *approx. periods in buffer*       integer (negative indicates error)
-   rate_numden                  *numerator, denominator*          tuple (integer (Hz), integer (Hz))
-   significant_bits             *significant bits in sample*      integer (negative indicates error)
-   is_batch                     *hw: double buffering*            boolean (True: hardware supported)
-   is_block_transfer            *hw: block transfer*              boolean (True: hardware supported)
-   is_double                    *hw: double buffering*            boolean (True: hardware supported)
-   is_half_duplex               *hw: half-duplex*                 boolean (True: hardware supported)
-   is_joint_duplex              *hw: joint-duplex*                boolean (True: hardware supported)
-   can_overrange                *hw: overrange detection*         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_resume                   *hw: resume*                      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():".
+   A small subset of properties reflects fixed parameters given by the
+   user, stored within alsaaudio. To distinguish them from properties
+   retrieved from ALSA when the call is made, they have their name
+   prefixed with **" (call value) "**.
 
+   Descriptions of properties which can be directly set during PCM object
+   instantiation carry the prefix "PCM():", followed by the respective
+   constructor parameter. Note that due to device limitations, the values
+   may deviate from those originally requested.
 
-.. method:: PCM.pcmtype()
+   Yet another set of properties cannot be set, and derives directly from
+   the hardware, possibly depending on other properties. Those properties'
+   descriptions are prefixed with "hw:" below.
+
+   ===========================  ====================================  ==================================================================
+        Key                      Description (Reference)               Type
+   ===========================  ====================================  ==================================================================
+   name                         PCM():device                          string
+   card_no                      *index of card*                       integer  (negative indicates device not associable with a card)
+   device_no                    *index of PCM device*                 integer
+   subdevice_no                 *index of PCM subdevice*              integer
+   state                        *name of PCM state*                   string
+   access_type                  *name of PCM access type*             string
+   (call value) type            PCM():type                            integer
+   (call value) type_name       PCM():type                            string
+   (call value) mode            PCM():mode                            integer
+   (call value) mode_name       PCM():mode                            string
+   format                       PCM():format                          integer
+   format_name                  PCM():format                          string
+   format_description           PCM():format                          string
+   subformat_name               *name of PCM subformat*               string
+   subformat_description        *description of subformat*            string
+   channels                     PCM():channels                        integer
+   rate                         PCM():rate                            integer (Hz)
+   period_time                  *period duration*                     integer (:math:`\mu s`)
+   period_size                  PCM():period_size                     integer (frames)
+   buffer_time                  *buffer time*                         integer (:math:`\mu s`) (negative indicates error)
+   buffer_size                  *buffer size*                         integer (frames) (negative indicates error)
+   get_periods                  *approx. periods in buffer*           integer (negative indicates error)
+   rate_numden                  *numerator, denominator*              tuple (integer (Hz), integer (Hz))
+   significant_bits             *significant bits in sample* [#tss]_  integer (negative indicates error)
+   nominal_bits                 *nominal bits in sample* [#tss]_      integer (negative indicates error)
+   physical_bits                *sample width in bits* [#tss]_        integer (negative indicates error)
+   is_batch                     *hw: double buffering*                boolean (True: hardware supported)
+   is_block_transfer            *hw: block transfer*                  boolean (True: hardware supported)
+   is_double                    *hw: double buffering*                boolean (True: hardware supported)
+   is_half_duplex               *hw: half-duplex*                     boolean (True: hardware supported)
+   is_joint_duplex              *hw: joint-duplex*                    boolean (True: hardware supported)
+   can_overrange                *hw: overrange detection*             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_resume                   *hw: resume*                          boolean (True: hardware supported)
+   can_sync_start               *hw: synchronized start*              boolean (True: hardware supported)
+   ===========================  ====================================  ==================================================================
+.. [#tss] More information in the :ref:`terminology section for sample size <term-sample-size>`
+
+..
+
+   The italicized descriptions give a summary of the "full" description
+   as can be found in the
+   `ALSA documentation <https://www.alsa-project.org/alsa-doc>`_.
+
+   *New in 0.9.1*
+
+.. method:: PCM.dumpinfo()
+
+   Dumps the PCM object's configured parameters to stdout.
+
+.. method:: PCM.pcmtype() -> int
 
    Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
    :const:`PCM_PLAYBACK`.
 
-.. method:: PCM.pcmmode()
+.. method:: PCM.pcmmode() -> int
 
    Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
    :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
 
-.. method:: PCM.cardname()
+.. method:: PCM.cardname() -> string
 
    Return the name of the sound card used by this PCM object.
 
@@ -267,37 +297,23 @@ PCM objects have the following methods:
    Returns a dictionary of supported format codes (integers) keyed by
    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`.
 
-.. method:: PCM.setrate(rate)
+.. method:: PCM.setrate(rate: int) -> int
 
    .. 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`.
 
-.. method:: PCM.setperiodsize(period)
+.. method:: PCM.setperiodsize(period: int) -> int
 
    .. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
 
-.. method:: PCM.info()
+.. method:: PCM.state() -> int
-
-   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*
-
-.. method:: PCM.dumpinfo()
-
-   Dumps the PCM object's configured parameters to stdout.
-
-.. method:: PCM.state()
 
    Returs the current state of the stream, which can be one of
    :const:`PCM_STATE_OPEN` (this should not actually happen),
@@ -312,54 +328,80 @@ PCM objects have the following methods:
 
    *New in 0.10*
 
-.. method:: PCM.read()
+.. method:: PCM.avail() -> int
+
+   For :const:`PCM_PLAYBACK` PCM objects, returns the number of writable
+   (that is, free) frames in the buffer.
+
+   For :const:`PCM_CAPTURE` PCM objects, returns the number of readable
+   (that is, filled) frames in the buffer.
+
+   An attempt to read/write more frames than indicated will block (in
+   :const:`PCM_NORMAL` mode) or fail and return zero (in
+   :const:`PCM_NONBLOCK` mode).
+
+   *New in 0.11*
+
+.. method:: PCM.read() -> tuple[int, bytes]
 
    In :const:`PCM_NORMAL` mode, this function blocks until a full period is
    available, and then returns a tuple (length,data) where *length* is
    the number of frames of captured data, and *data* is the captured
-   sound frames as a string. The length of the returned data will be 
+   sound frames as a string. The length of the returned data will be
    periodsize\*framesize bytes.
 
    In :const:`PCM_NONBLOCK` mode, the call will not block, but will return
    ``(0,'')`` if no new period has become available since the last
    call to read.
 
-   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
    multiple of the frame size, and *should* be exactly the size of a
    period. If less than 'period size' frames are provided, the actual
    playout will not happen until more data is written.
 
-   If the device is not in :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
    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
    in the kernel, and playout will continue afterwards. Make sure that the
    stream is drained before discarding the PCM handle.
 
-.. method:: PCM.pause([enable=True])
+.. method:: PCM.pause([enable: int = True]) -> int
 
    If *enable* is :const:`True`, playback or capture is paused.
    Otherwise, playback/capture is resumed.
 
-.. method:: PCM.drop()
+.. method:: PCM.drop() -> int
 
    Stop the stream and drop residual buffered frames.
 
    *New in 0.9*
 
-.. method:: PCM.drain()
+.. method:: PCM.drain() -> int
 
    For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
    and then stop the stream. In :const:`PCM_NORMAL` mode,
@@ -369,44 +411,53 @@ PCM objects have the following methods:
 
    *New in 0.10*
 
-.. method:: PCM.close()
+.. 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.
 
-.. method:: PCM.polldescriptors()
+.. method:: PCM.polldescriptors() -> list[tuple[int, int]]
 
    Returns a list of tuples of *(file descriptor, eventmask)* that can be
    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.
 
-.. method:: PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE])
+.. method:: PCM.polldescriptors_revents(descriptors: list[tuple[int, int]]) -> int
+
+   Processes the descriptor list returned by :func:`polldescriptors` after
+   using it with *select.poll*, and returns a single *eventmask* value that
+   is meaningful for deciding whether :func:`read` or :func:`write` should
+   be called.
+
+   *New in 0.11*
+
+.. method:: PCM.set_tstamp_mode([mode: int = PCM_TSTAMP_ENABLE])
 
    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`.
 
-.. method:: PCM.get_tstamp_mode()
+.. method:: PCM.get_tstamp_mode() -> int
 
    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`.
 
-.. method:: PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY])
+.. method:: PCM.set_tstamp_type([type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY]) -> None
 
    Set the ALSA timestamp mode on the device. The type argument
    can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
    :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 value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
    :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)*.
 
@@ -435,11 +486,11 @@ PCM objects have the following methods:
 
 **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.
 
 If too little data is written to the device, it will underrun, and
-ugly clicking sounds will occur. Conversely, of too much data is
+ugly clicking sounds will occur. Conversely, if too much data is
 written to the device, the write function will either block
 (:const:`PCM_NORMAL` mode) or return zero (:const:`PCM_NONBLOCK` mode).
 
@@ -468,12 +519,12 @@ Mixer Objects
 
 Mixer objects provides access to the ALSA mixer API.
 
-.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default')
+.. class:: Mixer(control: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> Mixer
 
    Arguments are:
 
    * *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
      ``'Master'`` - other common controls may be ``'Master Mono'``, ``'PCM'``,
      ``'Line'``, etc.
@@ -483,7 +534,7 @@ Mixer objects provides access to the ALSA mixer API.
    * *cardindex* - specifies which card should be used. If this argument
      is given, the device name is constructed like this: 'hw:*cardindex*' and
      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
      value is ``'default'``.
@@ -495,20 +546,20 @@ Mixer objects provides access to the ALSA mixer API.
 
 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
 
-.. method:: Mixer.mixer()
+.. method:: Mixer.mixer() -> str
 
    Return the name of the specific mixer controlled by this object, For example
    ``'Master'`` or ``'PCM'``
 
-.. method:: Mixer.mixerid()
+.. method:: Mixer.mixerid() -> int
 
    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.
    Possible values in this list are:
@@ -520,7 +571,7 @@ Mixer objects have the following methods:
    'Joined Mute'           This mixer can mute all channels at the same time
    'Playback Mute'         This mixer can mute the playback output
    'Joined Playback Mute'  Mute playback for all channels at the same time}
-   'Capture Mute'          Mute sound capture 
+   'Capture Mute'          Mute sound capture
    'Joined Capture Mute'   Mute sound capture for all channels at a time}
    'Capture Exclusive'     Not quite sure what this is
    ======================  ================
@@ -528,7 +579,7 @@ Mixer objects have the following methods:
    To manipulate these switches use the :meth:`setrec` or
    :meth:`setmute` methods
 
-.. method:: Mixer.volumecap()
+.. method:: Mixer.volumecap() -> int
 
    Returns a list of the volume control capabilities of this
    mixer. Possible values in the list are:
@@ -544,7 +595,7 @@ Mixer objects have the following methods:
    '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
    items available.
@@ -570,13 +621,13 @@ Mixer objects have the following methods:
    This method will return an empty tuple if the mixer is not an  enumerated
    control.
 
-.. method:: Mixer.setenum(index)
+.. method:: Mixer.setenum(index: int) -> None
 
    For enumerated controls, sets the currently selected item.
    *index* is an index into the list of available enumerated items returned
    by :func:`getenum`.
 
-.. method:: Mixer.getrange(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_RAW)
+.. method:: Mixer.getrange(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]
 
    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
@@ -590,7 +641,7 @@ Mixer objects have the following methods:
    The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
    :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
    elements are integers whose meaning is determined by the *units* argument.
@@ -603,7 +654,7 @@ Mixer objects have the following methods:
    The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
    :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) -> None
 
    Change the current volume settings for this mixer. The *volume* argument
    is an integer whose meaning is determined by the *units* argument.
@@ -620,14 +671,14 @@ Mixer objects have the following methods:
    The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
    :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.
    0 means not muted, 1 means muted.
 
    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
    muted, or 1 for muted.
@@ -637,14 +688,14 @@ Mixer objects have the following methods:
 
    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.
    0 means not recording, 1 means recording.
 
    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
    is either 0 for no capture, or 1 for capture.
@@ -654,21 +705,21 @@ Mixer objects have the following methods:
 
    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
    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.
 
-.. method:: Mixer.handleevents()
+.. method:: Mixer.handleevents() -> int
 
    Acknowledge events on the :func:`polldescriptors` file descriptors
    to prevent subsequent polls from returning the same events again.
    Returns the number of events that were acknowledged.
 
-.. method:: Mixer.close()
+.. method:: Mixer.close() -> None
 
    Closes the Mixer device.
 
@@ -707,7 +758,7 @@ The following example are provided:
 * `playbacktest.py`
 * `mixertest.py`
 
-All examples (except `mixertest.py`) accept the commandline option 
+All examples (except `mixertest.py`) accept the commandline option
 *-c <cardname>*.
 
 To determine a valid card name, use the commandline ALSA player::
@@ -722,12 +773,12 @@ or::
    >>> alsaaudio.pcms()
 
 mixertest.py accepts the commandline options *-d <device>* and
-*-c <cardindex>*. 
+*-c <cardindex>*.
 
 playwav.py
 ~~~~~~~~~~
 
-**playwav.py** plays a wav file. 
+**playwav.py** plays a wav file.
 
 To test PCM playback (on your default soundcard), run::
 
@@ -775,7 +826,7 @@ The output might look like this::
      'Mix'
      '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::
 
   $ ./mixertest.py Master
@@ -784,7 +835,7 @@ that control; for example::
   Channel 0 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::
 
   $ ./mixertest.py Master mute

doc/pyalsaaudio.rst

@@ -81,7 +81,7 @@ and need the ALSA headers for compilation.  Verify that you have
 
 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 
+may need to install the ALSA patches yourself - although most distributions
 ship with ALSA kernels.
 
 To install, execute the following:  ---   ::

doc/terminology.rst

@@ -6,7 +6,7 @@ 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*. 
+   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
@@ -22,8 +22,8 @@ Sample
    loudest signal that can be reproduced.
 
 Frame
-   A frame consists of exactly one sample per channel. If there is only one 
+   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 
+   channel (Mono sound) a frame is simply a single sample. If the sound is
    stereo, each frame consists of two samples, etc.
 
 Frame size
@@ -33,7 +33,7 @@ Frame size
    is 48 bytes.
 
 Rate
-   PCM sound consists of a flow of sound frames. The sound rate controls how 
+   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.
 
@@ -85,6 +85,21 @@ Period size
    each write should contain exactly 32 frames of sound data, and each
    read will return either 32 frames of data or nothing at all.
 
+.. _term-sample-size:
+
+Sample size
+   Each sample takes *physical_bits* of space. *nominal_bits* tells
+   how many least significant bits are used. This is the bit depth
+   in the format name and sometimes called just *sample bits* or
+   *format width*. *significant_bits* tells how many most significant
+   bits of the *nominal_bits* are used by the sample. This can be thought
+   of as the *sample resolution*. This is visualized as follows::
+
+    MSB |00000000 XXXXXXXX XXXXXXXX 00000000| LSB
+                  |--significant--|
+                  |---------nominal---------|
+        |-------------physical--------------|
+
 Once you understand these concepts, you will be ready to use the PCM API. Read
 on.
 

isine.py

@@ -7,6 +7,7 @@
 from __future__ import print_function
 
 import sys
+import time
 from threading import Thread
 from multiprocessing import Queue
 
@@ -15,14 +16,15 @@ if sys.version_info[0] < 3:
 else:
     from queue import Empty
 
-from math import pi, sin
+from math import pi, sin, ceil
 import struct
+import itertools
 import alsaaudio
 
 sampling_rate = 48000
 
 format = alsaaudio.PCM_FORMAT_S16_LE
-framesize = 2 # bytes per frame for the values above
+pack_format = 'h'  # short int, matching S16
 channels = 2
 
 def nearest_frequency(frequency):
@@ -34,28 +36,28 @@ def generate(frequency, duration = 0.125):
     # generate a buffer with a sine wave of `frequency`
     # that is approximately `duration` seconds long
 
-    # the buffersize we approximately want
-    target_size = int(sampling_rate * channels * duration)
-
     # the length of a full sine wave at the frequency
     cycle_size = int(sampling_rate / frequency)
 
-    # number of full cycles we can fit into target_size
+    # number of full cycles we can fit into the duration
-    factor = int(target_size / cycle_size)
+    factor = int(ceil(duration * frequency))
+
+    # total number of frames
+    size = cycle_size * factor
 
-    size = max(int(cycle_size * factor), 1)
-    
     sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \
              for i in range(size)]
-             
+
-    return struct.pack('%dh' % size, *sine)
+    if channels > 1:
-                                                                                  
+        sine = list(itertools.chain.from_iterable(itertools.repeat(x, channels) for x in sine))
+
+    return struct.pack(str(size * channels) + pack_format, *sine)
+
 
 class SinePlayer(Thread):
-    
+
     def __init__(self, frequency = 440.0):
-        Thread.__init__(self)
+        Thread.__init__(self, daemon=True)
-        self.setDaemon(True)
         self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
         self.queue = Queue()
         self.change(frequency)
@@ -73,22 +75,23 @@ class SinePlayer(Thread):
 
         buf = generate(f)
         self.queue.put(buf)
-                        
+
     def run(self):
         buffer = None
         while True:
             try:
                 buffer = self.queue.get(False)
-                self.device.setperiodsize(int(len(buffer) / framesize))
-                self.device.write(buffer)
             except Empty:
-                if buffer:
+                pass
-                    self.device.write(buffer)
+            if buffer:
-                
+                if self.device.write(buffer) < 0:
+                    print("Playback buffer underrun! Continuing nonetheless ...")
+
 
 isine = SinePlayer()
 isine.start()
 
-def change(f):
+time.sleep(1)
-    isine.change(f)
+isine.change(1000)
-    
+time.sleep(1)
+

loopback.py

@@ -0,0 +1,403 @@
+#!/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 = None
+		if run_after_stop:
+			self.run_after_stop = run_after_stop.split(' ')
+
+		self.run_before_start = None
+		if run_before_start:
+			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
+		self.volume = None
+
+	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()

mixertest.py

@@ -72,8 +72,13 @@ def show_mixer(name, kwargs):
     volumes = mixer.getvolume()
     volumes_dB = mixer.getvolume(units=alsaaudio.VOLUME_UNITS_DB)
     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:
         mutes = mixer.getmute()
         for i in range(len(mutes)):
@@ -113,7 +118,7 @@ def set_mixer(name, args, kwargs):
             mixer.setmute(1, channel)
         else:
             mixer.setmute(0, channel)
-        
+
     elif args in ['rec','unrec']:
         # Enable/disable recording
         if args == 'rec':

playbacktest.py

@@ -47,7 +47,8 @@ if __name__ == '__main__':
     # Read data from stdin
     data = f.read(320)
     while data:
-        out.write(data)
+        if out.write(data) < 0:
+            print("Playback buffer underrun! Continuing nonetheless ...")
         data = f.read(320)
     out.close()
-        
+

playwav.py

@@ -39,7 +39,8 @@ def play(device, f):
 	data = f.readframes(periodsize)
 	while data:
 		# Read data from stdin
-		device.write(data)
+		if device.write(data) < 0:
+			print("Playback buffer underrun! Continuing nonetheless ...")
 		data = f.readframes(periodsize)
 
 

recordtest.py

@@ -40,7 +40,7 @@ if __name__ == '__main__':
 
 	f = open(args[0], 'wb')
 
-	# Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz 
+	# Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz
 	# and 16 bit little endian samples
 	# The period size controls the internal number of frames per period.
 	# The significance of this parameter is documented in the ALSA api.
@@ -49,8 +49,8 @@ if __name__ == '__main__':
 	# This means that the reads below will return either 320 bytes of data
 	# or 0 bytes of data. The latter is possible because we are in nonblocking
 	# mode.
-	inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, 
+	inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK,
-		channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, 
+		channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE,
 		periodsize=160, device=device)
 
 	loops = 1000000
@@ -59,6 +59,8 @@ if __name__ == '__main__':
 		# Read data from device
 		l, data = inp.read()
 
-		if l:
+		if l < 0:
+			print("Capture buffer overrun! Continuing nonetheless ...")
+		elif l:
 			f.write(data)
 			time.sleep(.001)

setup.py

@@ -8,7 +8,7 @@ from setuptools import setup
 from setuptools.extension import Extension
 from sys import version
 
-pyalsa_version = '0.10.0'
+pyalsa_version = '0.10.1'
 
 if __name__ == '__main__':
     setup(
@@ -29,12 +29,12 @@ if __name__ == '__main__':
             'License :: OSI Approved :: Python Software Foundation License',
             'Operating System :: POSIX :: Linux',
             'Programming Language :: Python :: 2',
-            'Programming Language :: Python :: 3',    
+            'Programming Language :: Python :: 3',
             'Topic :: Multimedia :: Sound/Audio',
             'Topic :: Multimedia :: Sound/Audio :: Mixers',
             'Topic :: Multimedia :: Sound/Audio :: Players',
             'Topic :: Multimedia :: Sound/Audio :: Capture/Recording',
             ],
-        ext_modules=[Extension('alsaaudio',['alsaaudio.c'], 
+        ext_modules=[Extension('alsaaudio',['alsaaudio.c'],
                                libraries=['asound'])]
     )

test.py

@@ -11,6 +11,7 @@
 import unittest
 import alsaaudio
 import warnings
+from contextlib import closing
 
 # we can't test read and write well - these are tested otherwise
 PCMMethods = [
@@ -20,7 +21,7 @@ PCMMethods = [
 ]
 
 PCMDeprecatedMethods = [
-	('setchannels', (2,)), 
+	('setchannels', (2,)),
 	('setrate', (44100,)),
 	('setformat', (alsaaudio.PCM_FORMAT_S8,)),
 	('setperiodsize', (320,))
@@ -49,10 +50,10 @@ class MixerTest(unittest.TestCase):
 
 	def testMixer(self):
 		"""Open the default Mixers and the Mixers on every card"""
-		
+
 		for c in alsaaudio.card_indexes():
 			mixers = alsaaudio.mixers(cardindex=c)
-			
+
 			for m in mixers:
 				mixer = alsaaudio.Mixer(m, cardindex=c)
 				mixer.close()
@@ -73,7 +74,7 @@ class MixerTest(unittest.TestCase):
 		mixer.close()
 
 	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"""
 
 		mixers = alsaaudio.mixers()
@@ -133,7 +134,7 @@ class PCMTest(unittest.TestCase):
 				pcm = alsaaudio.PCM(card='default')
 			except alsaaudio.ALSAAudioError:
 				pass
-				
+
 			# Verify we got a DepreciationWarning
 			self.assertEqual(len(w), 1, "PCM(card='default') expected a warning" )
 			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.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__':
 	unittest.main()