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Added Freddy Spierenburgs patch for alsamixer_getrange and alsamixer_getenum.

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git-svn-id: svn://svn.code.sf.net/p/pyalsaaudio/code/trunk@15 ec2f30ec-7544-0410-870e-f70ca00c83f0
This commit is contained in:
larsimmisch
2008-01-24 13:36:56 +00:00
parent 2cf24ef81b
commit ba03ad3c75
3 changed files with 190 additions and 62 deletions
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214
alsaaudio.c
View File

@@ -1,3 +1,5 @@
/* -*- Mode: C; c-basic-offset: 2; indent-tabs-mode: nil -*- */
/*
* alsaaudio -- Python interface to ALSA (Advanced Linux Sound Architecture).
* The standard audio API for Linux since kernel 2.6
@@ -5,7 +7,7 @@
* Contributed by Unispeed A/S (http://www.unispeed.com)
* Author: Casper Wilstup (cwi@unispeed.dk)
*
* Bug fixes by Lars Immisch <lars@ibp.de>
* Bug fixes and maintenance by Lars Immisch <lars@ibp.de>
*
* License: Python Software Foundation License
*
@@ -16,14 +18,14 @@
#include <stdio.h>
PyDoc_STRVAR(alsaaudio_module_doc,
"This modules provides support for the ALSA audio API.\n"
"\n"
"To control the PCM device, use the PCM class, Mixers\n"
"are controlled using the Mixer class.\n"
"\n"
"The following functions are also provided:\n"
"mixers() -- Return a list of available mixer names\n"
);
"This modules provides support for the ALSA audio API.\n"
"\n"
"To control the PCM device, use the PCM class, Mixers\n"
"are controlled using the Mixer class.\n"
"\n"
"The following functions are also provided:\n"
"mixers() -- Return a list of available mixer names\n"
);
typedef struct {
PyObject_HEAD;
@@ -85,7 +87,8 @@ static int alsapcm_setup(alsapcm_t *self) {
snd_pcm_close(self->handle);
self->handle = 0;
}
res = snd_pcm_open(&(self->handle),self->cardname,self->pcmtype,self->pcmmode);
res = snd_pcm_open(&(self->handle),self->cardname,self->pcmtype,
self->pcmmode);
if (res < 0) return res;
/* Allocate a hwparam structure, and fill it in with configuration space */
@@ -95,12 +98,14 @@ static int alsapcm_setup(alsapcm_t *self) {
/* Fill it in with default values. */
snd_pcm_hw_params_any(self->handle, hwparams);
snd_pcm_hw_params_set_access(self->handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_access(self->handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(self->handle, hwparams, self->format);
snd_pcm_hw_params_set_channels(self->handle, hwparams, self->channels);
dir = 0;
snd_pcm_hw_params_set_rate(self->handle, hwparams, self->rate, dir);
snd_pcm_hw_params_set_period_size(self->handle, hwparams, self->periodsize, dir);
snd_pcm_hw_params_set_period_size(self->handle, hwparams, self->periodsize,
dir);
snd_pcm_hw_params_set_periods(self->handle,hwparams,4,0);
/* Write it to the device */
@@ -114,7 +119,8 @@ static int alsapcm_setup(alsapcm_t *self) {
snd_pcm_hw_params_get_format(hwparams,&fmt); self->format = fmt;
snd_pcm_hw_params_get_channels(hwparams,&val); self->channels = val;
snd_pcm_hw_params_get_rate(hwparams,&val,&dir); self->rate = val;
snd_pcm_hw_params_get_period_size(hwparams,&frames,&dir); self->periodsize = (int) frames;
snd_pcm_hw_params_get_period_size(hwparams,&frames,&dir);
self->periodsize = (int) frames;
self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8;
return res;
@@ -127,11 +133,15 @@ alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
int pcmtype=0;
int pcmmode=0;
char *cardname = "default";
if (!PyArg_ParseTuple(args,"|iis",&pcmtype,&pcmmode,&cardname)) return NULL;
if (!(self = (alsapcm_t *)PyObject_New(alsapcm_t, &ALSAPCMType))) return NULL;
if (!PyArg_ParseTuple(args,"|iis",&pcmtype,&pcmmode,&cardname))
return NULL;
if (!(self = (alsapcm_t *)PyObject_New(alsapcm_t, &ALSAPCMType)))
return NULL;
if (pcmtype != SND_PCM_STREAM_PLAYBACK && pcmtype != SND_PCM_STREAM_CAPTURE) {
PyErr_SetString(ALSAAudioError, "PCM type must be PCM_PLAYBACK (0) or PCM_CAPTURE (1)");
if (pcmtype != SND_PCM_STREAM_PLAYBACK && pcmtype != SND_PCM_STREAM_CAPTURE)
{
PyErr_SetString(ALSAAudioError, "PCM type must be PCM_PLAYBACK (0) "
"or PCM_CAPTURE (1)");
return NULL;
}
if (pcmmode < 0 || pcmmode > SND_PCM_ASYNC) {
@@ -191,13 +201,13 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) {
snd_pcm_hw_params_get_format(hwparams, &fmt);
printf("format = '%s' (%s)\n",
snd_pcm_format_name(fmt),
snd_pcm_format_description(fmt));
snd_pcm_format_name(fmt),
snd_pcm_format_description(fmt));
snd_pcm_hw_params_get_subformat(hwparams, (snd_pcm_subformat_t *)&val);
printf("subformat = '%s' (%s)\n",
snd_pcm_subformat_name((snd_pcm_subformat_t)val),
snd_pcm_subformat_description((snd_pcm_subformat_t)val));
snd_pcm_subformat_name((snd_pcm_subformat_t)val),
snd_pcm_subformat_description((snd_pcm_subformat_t)val));
snd_pcm_hw_params_get_channels(hwparams, &val);
printf("channels = %d\n", val);
@@ -345,7 +355,8 @@ alsapcm_read(alsapcm_t *self, PyObject *args) {
char buffer[8000];
if (self->framesize * self->periodsize > 8000) {
PyErr_SetString(ALSAAudioError,"Capture data too large. Try decreasing period size");
PyErr_SetString(ALSAAudioError,"Capture data too large. "
"Try decreasing period size");
return NULL;
}
@@ -366,11 +377,11 @@ alsapcm_read(alsapcm_t *self, PyObject *args) {
if (res != -EPIPE)
{
if (res == -EAGAIN) {
res = 0;
}
else if (res < 0) {
PyErr_SetString(ALSAAudioError,snd_strerror(res));
return NULL;
res = 0;
}
else if (res < 0) {
PyErr_SetString(ALSAAudioError,snd_strerror(res));
return NULL;
}
}
@@ -383,7 +394,8 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) {
int res;
if (!PyArg_ParseTuple(args,"s#",&data,&datalen)) return NULL;
if (datalen%self->framesize) {
PyErr_SetString(ALSAAudioError,"Data size must be a multiple of framesize");
PyErr_SetString(ALSAAudioError,
"Data size must be a multiple of framesize");
return NULL;
}
@@ -392,8 +404,8 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) {
if (res == -EPIPE) {
/* EPIPE means underrun */
res = snd_pcm_recover(self->handle, res, 1);
if (res >= 0)
res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
if (res >= 0)
res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
}
Py_END_ALLOW_THREADS
@@ -520,7 +532,7 @@ alsamixer_list(PyObject *self, PyObject *args) {
PyObject *result = PyList_New(0);
if (!PyArg_ParseTuple(args,"|s",&cardname)) return NULL;
snd_mixer_selem_id_alloca(&sid);
err = alsamixer_gethandle(cardname,&handle);
if (err < 0) {
@@ -528,7 +540,9 @@ alsamixer_list(PyObject *self, PyObject *args) {
snd_mixer_close(handle);
return NULL;
}
for (elem = snd_mixer_first_elem(handle); elem; elem = snd_mixer_elem_next(elem)) {
for (elem = snd_mixer_first_elem(handle); elem;
elem = snd_mixer_elem_next(elem))
{
PyObject *mixer;
snd_mixer_selem_get_id(elem, sid);
mixer = PyString_FromString(snd_mixer_selem_id_get_name(sid));
@@ -560,8 +574,10 @@ alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
snd_mixer_elem_t *elem;
int channel;
if (!PyArg_ParseTuple(args,"|sis",&control,&id,&cardname)) return NULL;
if (!(self = (alsamixer_t *)PyObject_New(alsamixer_t, &ALSAMixerType))) return NULL;
if (!PyArg_ParseTuple(args,"|sis",&control,&id,&cardname))
return NULL;
if (!(self = (alsamixer_t *)PyObject_New(alsamixer_t, &ALSAMixerType)))
return NULL;
err = alsamixer_gethandle(cardname,&self->handle);
if (err<0) {
@@ -575,7 +591,8 @@ alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
elem = alsamixer_find_elem(self->handle,control,id);
if (!elem) {
char errtext[128];
sprintf(errtext,"Unable to find mixer control '%s',%i",self->controlname,self->controlid);
sprintf(errtext,"Unable to find mixer control '%s',%i",self->controlname,
self->controlid);
snd_mixer_close(self->handle);
PyErr_SetString(ALSAAudioError,errtext);
return NULL;
@@ -584,51 +601,60 @@ alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) {
self->volume_cap = self->switch_cap = 0;
if (snd_mixer_selem_has_common_volume(elem)) {
self->volume_cap |= MIXER_CAP_VOLUME;
if (snd_mixer_selem_has_playback_volume_joined(elem)) self->volume_cap |= MIXER_CAP_VOLUME_JOINED;
if (snd_mixer_selem_has_playback_volume_joined(elem))
self->volume_cap |= MIXER_CAP_VOLUME_JOINED;
}
else {
if (snd_mixer_selem_has_playback_volume(elem)) {
self->volume_cap |= MIXER_CAP_PVOLUME;
if (snd_mixer_selem_has_playback_volume_joined(elem)) self->volume_cap |= MIXER_CAP_PVOLUME_JOINED;
if (snd_mixer_selem_has_playback_volume_joined(elem))
self->volume_cap |= MIXER_CAP_PVOLUME_JOINED;
}
if (snd_mixer_selem_has_capture_volume(elem)) {
self->volume_cap |= MIXER_CAP_CVOLUME;
if (snd_mixer_selem_has_capture_volume_joined(elem)) self->volume_cap |= MIXER_CAP_CVOLUME_JOINED;
if (snd_mixer_selem_has_capture_volume_joined(elem))
self->volume_cap |= MIXER_CAP_CVOLUME_JOINED;
}
}
if (snd_mixer_selem_has_common_switch(elem)) {
self->switch_cap |= MIXER_CAP_SWITCH;
if (snd_mixer_selem_has_playback_switch_joined(elem)) self->switch_cap |= MIXER_CAP_SWITCH_JOINED;
if (snd_mixer_selem_has_playback_switch_joined(elem))
self->switch_cap |= MIXER_CAP_SWITCH_JOINED;
}
else {
if (snd_mixer_selem_has_playback_switch(elem)) {
self->switch_cap |= MIXER_CAP_PSWITCH;
if (snd_mixer_selem_has_playback_switch_joined(elem)) self->switch_cap |= MIXER_CAP_PSWITCH_JOINED;
if (snd_mixer_selem_has_playback_switch_joined(elem))
self->switch_cap |= MIXER_CAP_PSWITCH_JOINED;
}
if (snd_mixer_selem_has_capture_switch(elem)) {
self->switch_cap |= MIXER_CAP_CSWITCH;
if (snd_mixer_selem_has_capture_switch_joined(elem)) self->switch_cap |= MIXER_CAP_CSWITCH_JOINED;
if (snd_mixer_selem_has_capture_switch_exclusive(elem)) self->switch_cap |= MIXER_CAP_CSWITCH_EXCLUSIVE;
if (snd_mixer_selem_has_capture_switch_joined(elem))
self->switch_cap |= MIXER_CAP_CSWITCH_JOINED;
if (snd_mixer_selem_has_capture_switch_exclusive(elem))
self->switch_cap |= MIXER_CAP_CSWITCH_EXCLUSIVE;
}
}
self->pchannels = 0;
if (self->volume_cap | MIXER_CAP_PVOLUME || self->switch_cap | MIXER_CAP_PSWITCH) {
if (self->volume_cap | MIXER_CAP_PVOLUME ||
self->switch_cap | MIXER_CAP_PSWITCH) {
if (snd_mixer_selem_is_playback_mono(elem)) self->pchannels = 1;
else {
for (channel=0; channel <= SND_MIXER_SCHN_LAST; channel++) {
if (snd_mixer_selem_has_playback_channel(elem, channel)) self->pchannels++;
else break;
if (snd_mixer_selem_has_playback_channel(elem, channel)) self->pchannels++;
else break;
}
}
}
self->cchannels = 0;
if (self->volume_cap | MIXER_CAP_CVOLUME || self->switch_cap | MIXER_CAP_CSWITCH) {
if (self->volume_cap | MIXER_CAP_CVOLUME ||
self->switch_cap | MIXER_CAP_CSWITCH) {
if (snd_mixer_selem_is_capture_mono(elem)) self->cchannels = 1;
else {
for (channel=0; channel <= SND_MIXER_SCHN_LAST; channel++) {
if (snd_mixer_selem_has_capture_channel(elem, channel)) self->cchannels++;
else break;
if (snd_mixer_selem_has_capture_channel(elem, channel)) self->cchannels++;
else break;
}
}
}
@@ -754,19 +780,91 @@ alsamixer_getvolume(alsamixer_t *self, PyObject *args) {
}
result = PyList_New(0);
for (channel = 0; channel <= SND_MIXER_SCHN_LAST; channel++) {
if (direction == 0 && snd_mixer_selem_has_playback_channel(elem, channel)) {
if (direction == 0 && snd_mixer_selem_has_playback_channel(elem, channel))
{
snd_mixer_selem_get_playback_volume(elem, channel, &ival);
PyList_Append(result,PyInt_FromLong(alsamixer_getpercentage(self->pmin,self->pmax,ival)));
PyList_Append(
result, PyInt_FromLong(alsamixer_getpercentage(self->pmin,
self->pmax, ival)));
}
else if (direction == 1 && snd_mixer_selem_has_capture_channel(elem, channel)
&& snd_mixer_selem_has_capture_volume(elem)) {
else if (direction == 1
&& snd_mixer_selem_has_capture_channel(elem, channel)
&& snd_mixer_selem_has_capture_volume(elem)) {
snd_mixer_selem_get_capture_volume(elem, channel, &ival);
PyList_Append(result,PyInt_FromLong(alsamixer_getpercentage(self->cmin,self->cmax,ival)));
PyList_Append(
result,PyInt_FromLong(alsamixer_getpercentage(self->cmin,
self->cmax, ival)));
}
}
return result;
}
static PyObject *
alsamixer_getrange(alsamixer_t *self, PyObject *args) {
snd_mixer_elem_t *elem;
int direction;
char *dirstr = 0;
PyObject *result;
if (!PyArg_ParseTuple(args,"|s",&dirstr)) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!dirstr) {
if (self->pchannels) direction = 0;
else direction = 1;
}
else if (strcasecmp(dirstr,"playback")==0) direction = 0;
else if (strcasecmp(dirstr,"capture")==0) direction = 1;
else {
PyErr_SetString(ALSAAudioError,"Invalid direction argument for direction");
return NULL;
}
result = PyList_New(0);
if (direction == 0 && snd_mixer_selem_has_playback_channel(elem, 0)) {
PyList_Append(result,PyInt_FromLong(self->pmin));
PyList_Append(result,PyInt_FromLong(self->pmax));
}
else if (direction == 1 && snd_mixer_selem_has_capture_channel(elem, 0)
&& snd_mixer_selem_has_capture_volume(elem)) {
PyList_Append(result,PyInt_FromLong(self->cmin));
PyList_Append(result,PyInt_FromLong(self->cmax));
}
return result;
}
static PyObject *
alsamixer_getenum(alsamixer_t *self, PyObject *args) {
snd_mixer_elem_t *elem;
unsigned int index;
char name[32];
int res;
PyObject *result;
if (!PyArg_ParseTuple(args,"")) return NULL;
elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid);
if (!snd_mixer_selem_is_enumerated(elem)) {
PyErr_SetString(ALSAAudioError,"Mixer is no enumerated control");
return NULL;
}
res=snd_mixer_selem_get_enum_item(elem, 0, &index);
if(res) {
PyErr_SetString(ALSAAudioError, snd_strerror(res));
return NULL;
}
res=snd_mixer_selem_get_enum_item_name(elem, index, sizeof(name)-1, name);
if(res) {
PyErr_SetString(ALSAAudioError, snd_strerror(res));
return NULL;
} else {
result = PyList_New(0);
PyList_Append(result,PyString_FromString(name));
}
return result;
}
static PyObject *
alsamixer_getmute(alsamixer_t *self, PyObject *args) {
snd_mixer_elem_t *elem;
@@ -839,7 +937,8 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args) {
else if (strcasecmp(dirstr,"playback")==0) direction = 0;
else if (strcasecmp(dirstr,"capture")==0) direction = 1;
else {
PyErr_SetString(ALSAAudioError,"Invalid direction argument. Use 'playback' or 'capture'");
PyErr_SetString(ALSAAudioError,"Invalid direction argument. Use "
"'playback' or 'capture'");
return NULL;
}
for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) {
@@ -849,8 +948,9 @@ alsamixer_setvolume(alsamixer_t *self, PyObject *args) {
snd_mixer_selem_set_playback_volume(elem, i, physvolume);
done++;
}
else if (direction == 1 && snd_mixer_selem_has_capture_channel(elem, channel)
&& snd_mixer_selem_has_capture_volume(elem)) {
else if (direction == 1
&& snd_mixer_selem_has_capture_channel(elem, channel)
&& snd_mixer_selem_has_capture_volume(elem)) {
physvolume = alsamixer_getphysvolume(self->cmin,self->cmax,volume);
snd_mixer_selem_set_capture_volume(elem, i, physvolume);
done++;
@@ -932,6 +1032,8 @@ static PyMethodDef alsamixer_methods[] = {
{"switchcap", (PyCFunction)alsamixer_switchcap, METH_VARARGS},
{"volumecap", (PyCFunction)alsamixer_volumecap, METH_VARARGS},
{"getvolume", (PyCFunction)alsamixer_getvolume, METH_VARARGS},
{"getrange", (PyCFunction)alsamixer_getrange, METH_VARARGS},
{"getenum", (PyCFunction)alsamixer_getenum, METH_VARARGS},
{"getmute", (PyCFunction)alsamixer_getmute, METH_VARARGS},
{"getrec", (PyCFunction)alsamixer_getrec, METH_VARARGS},
{"setvolume", (PyCFunction)alsamixer_setvolume, METH_VARARGS},

28
doc/mixer-objects.html
View File

@@ -176,7 +176,23 @@ the list are:
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-22' xml:id='l2h-22' class="method">getvolume</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-22' xml:id='l2h-22' class="method">getrange</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Return the volume range of the ALSA mixer controlled by this object.
<p>
The optional <var>direction</var> argument can be either 'playback' or 'capture',
which is relevant if the mixer can control both playback and capture volume.
The default value is 'playback' if the mixer has this capability, otherwise
'capture'
<p>
</dl>
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-23' xml:id='l2h-23' class="method">getvolume</tt></b>(</nobr></td>
<td><var></var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Returns a list with the current volume settings for each channel. The list elements
@@ -192,7 +208,7 @@ if the mixer has this capability, otherwise 'capture'
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-23' xml:id='l2h-23' class="method">getmute</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-24' xml:id='l2h-24' class="method">getmute</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted.
@@ -203,7 +219,7 @@ This method will fail if the mixer has no playback switch capabilities.
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-24' xml:id='l2h-24' class="method">getrec</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-25' xml:id='l2h-25' class="method">getrec</tt></b>(</nobr></td>
<td><var></var>)</td></tr></table></dt>
<dd>
Return a list indicating the current record mute setting for each channel. 0 means not recording, 1
@@ -215,7 +231,7 @@ This method will fail if the mixer has no capture switch capabilities.
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-25' xml:id='l2h-25' class="method">setvolume</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-26' xml:id='l2h-26' class="method">setvolume</tt></b>(</nobr></td>
<td><var>volume,</var><big>[</big><var>channel</var><big>]</big><var>,</var><big>[</big><var>direction</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Change the current volume settings for this mixer. The <var>volume</var> argument controls
@@ -233,7 +249,7 @@ if changed. The default is 'playback' if the mixer has this capability, otherwis
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-26' xml:id='l2h-26' class="method">setmute</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-27' xml:id='l2h-27' class="method">setmute</tt></b>(</nobr></td>
<td><var>mute, </var><big>[</big><var>channel</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Sets the mute flag to a new value. The <var>mute</var> argument is either 0 for not muted, or 1 for muted.
@@ -248,7 +264,7 @@ This method will fail if the mixer has no playback mute capabilities
<p>
<dl><dt><table cellpadding="0" cellspacing="0"><tr valign="baseline">
<td><nobr><b><tt id='l2h-27' xml:id='l2h-27' class="method">setrec</tt></b>(</nobr></td>
<td><nobr><b><tt id='l2h-28' xml:id='l2h-28' class="method">setrec</tt></b>(</nobr></td>
<td><var>capture,</var><big>[</big><var>channel</var><big>]</big><var></var>)</td></tr></table></dt>
<dd>
Sets the capture mute flag to a new value. The <var>capture</var> argument is either 0 for no capture,

10
doc/src/libalsaaudio.tex
View File

@@ -315,6 +315,16 @@ the list are:
\end{methoddesc}
\begin{methoddesc}[Mixer]{getrange}{\optional{direction}}
Return the volume range of the ALSA mixer controlled by this object.
The optional \var{direction} argument can be either 'playback' or 'capture',
which is relevant if the mixer can control both playback and capture volume.
The default value is 'playback' if the mixer has this capability, otherwise
'capture'
\end{methoddesc}
\begin{methoddesc}[Mixer]{getvolume}{\optional{direction}}
Returns a list with the current volume settings for each channel. The list elements
are integer percentages.