From 2876869020584497e61a340d4ca3c4c8a8cbde7f Mon Sep 17 00:00:00 2001 From: Walter Bender Date: Sat, 21 Feb 2009 12:43:42 +0000 Subject: undoing pyalsaaudio patch --- diff --git a/activity/activity.info b/activity/activity.info index 4c268d2..8097efe 100644 --- a/activity/activity.info +++ b/activity/activity.info @@ -3,7 +3,7 @@ name = Turtle Art activity_version = 36 license = MIT bundle_id = org.laptop.TurtleArtActivity -exec = wrapper sugar-activity TurtleArtActivity.TurtleArtActivity +exec = sugar-activity TurtleArtActivity.TurtleArtActivity icon = activity-turtleart show_launcher = yes update_url = http://wiki.laptop.org/go/Activities/G1G1 diff --git a/arch_src/Makefile b/arch_src/Makefile deleted file mode 100644 index 2f915d5..0000000 --- a/arch_src/Makefile +++ /dev/null @@ -1,7 +0,0 @@ - -all: - cd pyalsaaudio-0.2 ; python setup.py build - -install: - cd pyalsaaudio-0.2 ; python setup.py install --prefix="${PREFIX}" - diff --git a/arch_src/pyalsaaudio-0.2/CHANGES b/arch_src/pyalsaaudio-0.2/CHANGES deleted file mode 100644 index 6ce8463..0000000 --- a/arch_src/pyalsaaudio-0.2/CHANGES +++ /dev/null @@ -1,8 +0,0 @@ -VERSION 0.1 -- Initial version - - -VERSION 0.2 -- Many bugfixes related to playback in particular -- Module documentation in the doc subdirectory - diff --git a/arch_src/pyalsaaudio-0.2/LICENSE b/arch_src/pyalsaaudio-0.2/LICENSE deleted file mode 100644 index 8aa69f7..0000000 --- a/arch_src/pyalsaaudio-0.2/LICENSE +++ /dev/null @@ -1,54 +0,0 @@ -PyAlsaAudio is released under the same conditions as Python itself. -The original wording of this license can be found below. - - -PSF LICENSE AGREEMENT FOR PYTHON 2.4 ------------------------------------- - -1. This LICENSE AGREEMENT is between the Python Software Foundation -("PSF"), and the Individual or Organization ("Licensee") accessing and -otherwise using Python 2.4 software in source or binary form and its -associated documentation. - -2. Subject to the terms and conditions of this License Agreement, PSF -hereby grants Licensee a nonexclusive, royalty-free, world-wide -license to reproduce, analyze, test, perform and/or display publicly, -prepare derivative works, distribute, and otherwise use Python 2.4 -alone or in any derivative version, provided, however, that PSF's -License Agreement and PSF's notice of copyright, i.e., "Copyright (c) -2001, 2002, 2003, 2004 Python Software Foundation; All Rights Reserved" -are retained in Python 2.4 alone or in any derivative version prepared -by Licensee. - -3. In the event Licensee prepares a derivative work that is based on -or incorporates Python 2.4 or any part thereof, and wants to make -the derivative work available to others as provided herein, then -Licensee hereby agrees to include in any such work a brief summary of -the changes made to Python 2.4. - -4. PSF is making Python 2.4 available to Licensee on an "AS IS" -basis. PSF MAKES NO REPRESENTATIONS OR WARRANTIES, EXPRESS OR -IMPLIED. BY WAY OF EXAMPLE, BUT NOT LIMITATION, PSF MAKES NO AND -DISCLAIMS ANY REPRESENTATION OR WARRANTY OF MERCHANTABILITY OR FITNESS -FOR ANY PARTICULAR PURPOSE OR THAT THE USE OF PYTHON 2.4 WILL NOT -INFRINGE ANY THIRD PARTY RIGHTS. - -5. PSF SHALL NOT BE LIABLE TO LICENSEE OR ANY OTHER USERS OF PYTHON -2.4 FOR ANY INCIDENTAL, SPECIAL, OR CONSEQUENTIAL DAMAGES OR LOSS AS -A RESULT OF MODIFYING, DISTRIBUTING, OR OTHERWISE USING PYTHON 2.4, -OR ANY DERIVATIVE THEREOF, EVEN IF ADVISED OF THE POSSIBILITY THEREOF. - -6. This License Agreement will automatically terminate upon a material -breach of its terms and conditions. - -7. Nothing in this License Agreement shall be deemed to create any -relationship of agency, partnership, or joint venture between PSF and -Licensee. This License Agreement does not grant permission to use PSF -trademarks or trade name in a trademark sense to endorse or promote -products or services of Licensee, or any third party. - -8. By copying, installing or otherwise using Python 2.4, Licensee -agrees to be bound by the terms and conditions of this License -Agreement. - - diff --git a/arch_src/pyalsaaudio-0.2/README b/arch_src/pyalsaaudio-0.2/README deleted file mode 100644 index 179cb79..0000000 --- a/arch_src/pyalsaaudio-0.2/README +++ /dev/null @@ -1,55 +0,0 @@ -PyAlsaAudio -=========== - -Author: Casper Wilstrup (cwi@unispeed.dk) - -This package contains wrappers for accessing the ALSA api from Python. It -is currently fairly complete for PCM devices. My next goal is to have -complete mixer supports as well. MIDI sequencer support is low on my -priority list, but volunteers are welcome. - -If you find bugs in the wrappers please notify me on email. Please -don't send bug reports regarding ALSA specifically. There are several -bugs in this api, and those should be reported to the ALSA team - not -me. - -This software is licensed under the PSF license - the same one used -by the majority of the python distribution. Basically you can use it -for anything you wish (even commercial purposes). There is no warranty -whatsoever. - - -Installation -============ - -Note: the wrappers link with the alsasound library alsa (from the alsa-lib -package). Verify that this is installed by looking for /usr/lib/libasound.so -before building. The libasound development files are also neccesary. On debian -and derivatives, this is achieved by installing the alsalib-dev package. - -Naturally you also need to use a kernel with proper ALSA -support. This is the default in Linux kernel 2.6 and later. If you are using -kernel version 2.4 you may need to install the ALSA patches yourself - although -most distributions ship with ALSA kernels. - -To install, execute the following: - $ python setup.py build - -And then as root: - # python setup.py install - - -Using the API -============= -There is a reasonably usefull API documentation included in the module -documentation, which can be found in the doc subdirectory of the source -distribution. - -There are also three example programs included with the source: -'playbacktest.py' which plays back raw sound data read from -stdin - -'recordtest.py' which captures sound from the microphone at writes -it raw to stdout. - -'mixertest.py' which can be used to manipulate the mixers diff --git a/arch_src/pyalsaaudio-0.2/TODO b/arch_src/pyalsaaudio-0.2/TODO deleted file mode 100644 index 710a8a3..0000000 --- a/arch_src/pyalsaaudio-0.2/TODO +++ /dev/null @@ -1,3 +0,0 @@ -- Better example code (aplay,arecord,amixer workalike for example) -- Implement MIDI/sequencer support. - diff --git a/arch_src/pyalsaaudio-0.2/alsaaudio.c b/arch_src/pyalsaaudio-0.2/alsaaudio.c deleted file mode 100644 index 1828148..0000000 --- a/arch_src/pyalsaaudio-0.2/alsaaudio.c +++ /dev/null @@ -1,1013 +0,0 @@ -/* - * alsaaudio -- Python interface to ALSA (Advanced Linux Sound Architecture). - * The standard audio API for Linux since kernel 2.6 - * - * Contributed by Unispeed A/S (http://www.unispeed.com) - * Author: Casper Wilstup (cwi@unispeed.dk) - * - * License: Python Software Foundation License - * - */ - -#include "Python.h" -#include -#include - -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" - ); - -typedef struct { - PyObject_HEAD; - int pcmtype; - int pcmmode; - char *cardname; - - snd_pcm_t *handle; - - // Configurable parameters - int channels; - int rate; - int format; - snd_pcm_uframes_t periodsize; - int framesize; - -} alsapcm_t; - -typedef struct { - PyObject_HEAD; - - /* Mixer identification */ - char *cardname; - char *controlname; - int controlid; - - /* Capabilities */ - unsigned int volume_cap; - unsigned int switch_cap; - unsigned int pchannels; - unsigned int cchannels; - - /* min and max values for playback and capture volumes */ - long pmin; - long pmax; - long cmin; - long cmax; - snd_mixer_t *handle; - -} alsamixer_t; - -static PyObject *ALSAAudioError; - - -/******************************************/ -/* PCM object wrapper */ -/******************************************/ - -static PyTypeObject ALSAPCMType; - -static int alsapcm_setup(alsapcm_t *self) { - int res,dir; - unsigned int val; - snd_pcm_uframes_t frames; - snd_pcm_hw_params_t *hwparams; - - if (self->handle) { - snd_pcm_close(self->handle); - self->handle = 0; - } - res = snd_pcm_open(&(self->handle),self->cardname,self->pcmtype,self->pcmmode); - if (res < 0) return res; - - /* Allocate a hwparam structure, and fill it in with configuration space */ - snd_pcm_hw_params_alloca(&hwparams); - res = snd_pcm_hw_params_any(self->handle, hwparams); - if (res < 0) return res; - - /* 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_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_periods(self->handle,hwparams,4,0); - - /* Write it to the device */ - res = snd_pcm_hw_params(self->handle, hwparams); - if (res) return res; - - /* Query current settings. These may differ from the requested values, - which should therefore be sync'ed with actual values */ - snd_pcm_hw_params_current(self->handle,hwparams); - - snd_pcm_hw_params_get_format(hwparams,&val); self->format = val; - snd_pcm_hw_params_get_channels(hwparams,&val); self->channels = val; - snd_pcm_hw_params_get_rate(hwparams,&val,&dir); self->rate = val; - snd_pcm_hw_params_get_period_size(hwparams,&frames,&dir); self->periodsize = (int) frames; - - self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8; - return res; -} - -static PyObject * -alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds) { - int res; - alsapcm_t *self; - int pcmtype=0; - int pcmmode=0; - char *cardname = "default"; - if (!PyArg_ParseTuple(args,"|iis",&pcmtype,&pcmmode,&cardname)) 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_CAPTUPE (1)"); - return NULL; - } - if (pcmmode < 0 || pcmmode > SND_PCM_ASYNC) { - PyErr_SetString(ALSAAudioError, "Invalid PCM mode"); - return NULL; - } - self->pcmtype = pcmtype; - self->pcmmode = pcmmode; - self->cardname = strdup(cardname); - - self->channels = 2; - self->rate = 44100; - self->format = SND_PCM_FORMAT_S16_LE; - self->periodsize = 32; - - self->handle = 0; - res = alsapcm_setup(self); - - if (res < 0) { - if (self->handle) { - snd_pcm_close(self->handle); - self->handle = 0; - } - PyErr_SetString(ALSAAudioError, snd_strerror(res)); - return NULL; - } - return (PyObject *)self; -} - -static void alsapcm_dealloc(alsapcm_t *self) { - if (self->handle) { - snd_pcm_drain(self->handle); - snd_pcm_close(self->handle); - } - free(self->cardname); - PyObject_Del(self); -} - -static PyObject * -alsapcm_dumpinfo(alsapcm_t *self, PyObject *args) { - unsigned int val,val2; - int dir; - snd_pcm_uframes_t frames; - snd_pcm_hw_params_t *hwparams; - snd_pcm_hw_params_alloca(&hwparams); - snd_pcm_hw_params_current(self->handle,hwparams); - - - if (!PyArg_ParseTuple(args,"")) return NULL; - - printf("PCM handle name = '%s'\n", snd_pcm_name(self->handle)); - printf("PCM state = %s\n", snd_pcm_state_name(snd_pcm_state(self->handle))); - - snd_pcm_hw_params_get_access(hwparams, (snd_pcm_access_t *) &val); - printf("access type = %s\n", snd_pcm_access_name((snd_pcm_access_t)val)); - - snd_pcm_hw_params_get_format(hwparams, &val); - printf("format = '%s' (%s)\n", - snd_pcm_format_name((snd_pcm_format_t)val), - snd_pcm_format_description((snd_pcm_format_t)val)); - - 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_hw_params_get_channels(hwparams, &val); - printf("channels = %d\n", val); - - snd_pcm_hw_params_get_rate(hwparams, &val, &dir); - printf("rate = %d bps\n", val); - - snd_pcm_hw_params_get_period_time(hwparams, &val, &dir); - printf("period time = %d us\n", val); - - snd_pcm_hw_params_get_period_size(hwparams, &frames, &dir); - printf("period size = %d frames\n", (int)frames); - - snd_pcm_hw_params_get_buffer_time(hwparams, &val, &dir); - printf("buffer time = %d us\n", val); - - snd_pcm_hw_params_get_buffer_size(hwparams, (snd_pcm_uframes_t *) &val); - printf("buffer size = %d frames\n", val); - - snd_pcm_hw_params_get_periods(hwparams, &val, &dir); - printf("periods per buffer = %d frames\n", val); - - snd_pcm_hw_params_get_rate_numden(hwparams, &val, &val2); - printf("exact rate = %d/%d bps\n", val, val2); - - val = snd_pcm_hw_params_get_sbits(hwparams); - printf("significant bits = %d\n", val); - - snd_pcm_hw_params_get_tick_time(hwparams, &val, &dir); - printf("tick time = %d us\n", val); - - val = snd_pcm_hw_params_is_batch(hwparams); - printf("is batch = %d\n", val); - - val = snd_pcm_hw_params_is_block_transfer(hwparams); - printf("is block transfer = %d\n", val); - - val = snd_pcm_hw_params_is_double(hwparams); - printf("is double = %d\n", val); - - val = snd_pcm_hw_params_is_half_duplex(hwparams); - printf("is half duplex = %d\n", val); - - val = snd_pcm_hw_params_is_joint_duplex(hwparams); - printf("is joint duplex = %d\n", val); - - val = snd_pcm_hw_params_can_overrange(hwparams); - printf("can overrange = %d\n", val); - - val = snd_pcm_hw_params_can_mmap_sample_resolution(hwparams); - printf("can mmap = %d\n", val); - - val = snd_pcm_hw_params_can_pause(hwparams); - printf("can pause = %d\n", val); - - val = snd_pcm_hw_params_can_resume(hwparams); - printf("can resume = %d\n", val); - - val = snd_pcm_hw_params_can_sync_start(hwparams); - printf("can sync start = %d\n", val); - - Py_INCREF(Py_None); - return Py_None; - -} - -static PyObject * -alsapcm_pcmtype(alsapcm_t *self, PyObject *args) { - if (!PyArg_ParseTuple(args,"")) return NULL; - return PyInt_FromLong(self->pcmtype); -} - -static PyObject * -alsapcm_pcmmode(alsapcm_t *self, PyObject *args) { - if (!PyArg_ParseTuple(args,"")) return NULL; - return PyInt_FromLong(self->pcmmode); -} - -static PyObject * -alsapcm_cardname(alsapcm_t *self, PyObject *args) { - if (!PyArg_ParseTuple(args,"")) return NULL; - return PyString_FromString(self->cardname); -} - -static PyObject * -alsapcm_setchannels(alsapcm_t *self, PyObject *args) { - int channels; - int res; - if (!PyArg_ParseTuple(args,"i",&channels)) return NULL; - self->channels = channels; - res = alsapcm_setup(self); - if (res < 0) { - PyErr_SetString(ALSAAudioError, snd_strerror(res)); - return NULL; - } - return PyInt_FromLong(self->channels); -} - -static PyObject * -alsapcm_setrate(alsapcm_t *self, PyObject *args) { - int rate; - int res; - if (!PyArg_ParseTuple(args,"i",&rate)) return NULL; - self->rate = rate; - res = alsapcm_setup(self); - if (res < 0) { - PyErr_SetString(ALSAAudioError, snd_strerror(res)); - return NULL; - } - return PyInt_FromLong(self->rate); -} - -static PyObject * -alsapcm_setformat(alsapcm_t *self, PyObject *args) { - int format; - int res; - if (!PyArg_ParseTuple(args,"i",&format)) return NULL; - self->format = format; - res = alsapcm_setup(self); - if (res < 0) { - PyErr_SetString(ALSAAudioError, snd_strerror(res)); - return NULL; - } - return PyInt_FromLong(self->format); -} - -static PyObject * -alsapcm_setperiodsize(alsapcm_t *self, PyObject *args) { - int periodsize; - int res; - if (!PyArg_ParseTuple(args,"i",&periodsize)) return NULL; - self->periodsize = periodsize; - res = alsapcm_setup(self); - if (res < 0) { - PyErr_SetString(ALSAAudioError, snd_strerror(res)); - return NULL; - } - return PyInt_FromLong(self->periodsize); -} - - -static PyObject * -alsapcm_read(alsapcm_t *self, PyObject *args) { - int res; - char buffer[8000]; - - if (self->framesize * self->periodsize > 8000) { - PyErr_SetString(ALSAAudioError,"Capture data too large. Try decreasing period size"); - return NULL; - } - - if (!PyArg_ParseTuple(args,"")) return NULL; - if (self->pcmtype != SND_PCM_STREAM_CAPTURE) { - PyErr_SetString(ALSAAudioError,"Cannot read from playback PCM"); - return NULL; - } - - res = snd_pcm_readi(self->handle, buffer, self->periodsize); - if (res == -EPIPE) { - /* EPIPE means overrun */ - snd_pcm_prepare(self->handle); - } - else if (res == -EAGAIN) { - res = 0; - } - else if (res < 0) { - PyErr_SetString(ALSAAudioError,snd_strerror(res)); - return NULL; - } - - return Py_BuildValue("is#",res,buffer,res*self->framesize); -} - -static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args) { - char *data; - int datalen; - int res; - if (!PyArg_ParseTuple(args,"s#",&data,&datalen)) return NULL; - if (datalen%self->framesize) { - PyErr_SetString(ALSAAudioError,"Data size must be a multiple of framesize"); - return NULL; - } - res = snd_pcm_writei(self->handle, data, datalen/self->framesize); - if (res == -EPIPE) { - /* EPIPE means underrun */ - snd_pcm_prepare(self->handle); - snd_pcm_writei(self->handle, data, datalen/self->framesize); - snd_pcm_writei(self->handle, data, datalen/self->framesize); - } - else if (res == -EAGAIN) { - return PyInt_FromLong(0); - } - else if (res < 0) { - PyErr_SetString(ALSAAudioError,snd_strerror(res)); - return NULL; - } - return PyInt_FromLong(res); -} - - -/* 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}, - {"setchannels", (PyCFunction)alsapcm_setchannels, METH_VARARGS}, - {"setrate", (PyCFunction)alsapcm_setrate, METH_VARARGS}, - {"setformat", (PyCFunction)alsapcm_setformat, METH_VARARGS}, - {"setperiodsize", (PyCFunction)alsapcm_setperiodsize, METH_VARARGS}, - - {"dumpinfo", (PyCFunction)alsapcm_dumpinfo, METH_VARARGS}, - - {"read", (PyCFunction)alsapcm_read, METH_VARARGS}, - {"write", (PyCFunction)alsapcm_write, METH_VARARGS}, - - {NULL, NULL} -}; - -static PyObject * -alsapcm_getattr(alsapcm_t *self, char *name) { - return Py_FindMethod(alsapcm_methods, (PyObject *)self, name); -} - -static PyTypeObject ALSAPCMType = { - PyObject_HEAD_INIT(&PyType_Type) - 0, /*ob_size*/ - "alsaaudio.pcm", /*tp_name*/ - sizeof(alsapcm_t), /*tp_basicsize*/ - 0, /*tp_itemsize*/ - /* methods */ - (destructor) alsapcm_dealloc, /*tp_dealloc*/ - 0, /*print*/ - (getattrfunc)alsapcm_getattr, /*tp_getattr*/ - 0, /*tp_setattr*/ - 0, /*tp_compare*/ - 0, /*tp_repr*/ - 0, /*tp_as_number*/ - 0, /*tp_as_sequence*/ - 0, /*tp_as_mapping*/ - 0, /*tp_hash*/ - 0, /*tp_call*/ - 0, /*tp_str*/ - 0, /*tp_getattro*/ - 0, /*tp_setattro*/ - 0, /*tp_as_buffer*/ - Py_TPFLAGS_DEFAULT, /*tp_flags*/ - "ALSA PCM device", /*tp_doc*/ -}; - - -/******************************************/ -/* Mixer object wrapper */ -/******************************************/ - -static PyTypeObject ALSAMixerType; - -#define MIXER_CAP_VOLUME 0x0001 -#define MIXER_CAP_VOLUME_JOINED 0x0002 -#define MIXER_CAP_PVOLUME 0x0004 -#define MIXER_CAP_PVOLUME_JOINED 0x0008 -#define MIXER_CAP_CVOLUME 0x0010 -#define MIXER_CAP_CVOLUME_JOINED 0x0020 - -#define MIXER_CAP_SWITCH 0x0001 -#define MIXER_CAP_SWITCH_JOINED 0x0002 -#define MIXER_CAP_PSWITCH 0x0004 -#define MIXER_CAP_PSWITCH_JOINED 0x0008 -#define MIXER_CAP_CSWITCH 0x0010 -#define MIXER_CAP_CSWITCH_JOINED 0x0020 -#define MIXER_CAP_CSWITCH_EXCLUSIVE 0x0040 - -#define MIXER_CHANNEL_ALL -1 - -int -alsamixer_gethandle(char *cardname, snd_mixer_t **handle) { - int err; - if ((err = snd_mixer_open(handle, 0)) < 0) return err; - if ((err = snd_mixer_attach(*handle, cardname)) < 0) return err; - if ((err = snd_mixer_selem_register(*handle, NULL, NULL)) < 0) return err; - if ((err = snd_mixer_load(*handle)) < 0) return err; - - return 0; -} - -static PyObject * -alsamixer_list(PyObject *self, PyObject *args) { - snd_mixer_t *handle; - snd_mixer_selem_id_t *sid; - snd_mixer_elem_t *elem; - int err; - char *cardname = "default"; - 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) { - PyErr_SetString(ALSAAudioError,snd_strerror(err)); - snd_mixer_close(handle); - return NULL; - } - 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)); - PyList_Append(result,mixer); - Py_DECREF(mixer); - } - snd_mixer_close(handle); - - return result; -} - -static snd_mixer_elem_t * -alsamixer_find_elem(snd_mixer_t *handle, char *control, int id) { - snd_mixer_selem_id_t *sid; - - snd_mixer_selem_id_alloca(&sid); - snd_mixer_selem_id_set_index(sid, id); - snd_mixer_selem_id_set_name(sid, control); - return snd_mixer_find_selem(handle, sid); -} - -static PyObject * -alsamixer_new(PyTypeObject *type, PyObject *args, PyObject *kwds) { - alsamixer_t *self; - int err; - char *cardname = "default"; - char *control = "Master"; - int id = 0; - 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; - - err = alsamixer_gethandle(cardname,&self->handle); - if (err<0) { - PyErr_SetString(ALSAAudioError,snd_strerror(err)); - return NULL; - } - self->cardname = strdup(cardname); - self->controlname = strdup(control); - self->controlid = id; - - 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); - snd_mixer_close(self->handle); - PyErr_SetString(ALSAAudioError,errtext); - return NULL; - } - /* Determine mixer capabilities */ - 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; - } - 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_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_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; - } - 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_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; - } - } - self->pchannels = 0; - 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; - } - } - } - self->cchannels = 0; - 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; - } - } - } - snd_mixer_selem_get_playback_volume_range(elem, &self->pmin, &self->pmax); - snd_mixer_selem_get_capture_volume_range(elem, &self->cmin, &self->cmax); - return (PyObject *)self; -} - -static void alsamixer_dealloc(alsamixer_t *self) { - if (self->handle) { - snd_mixer_close(self->handle); - free(self->cardname); - free(self->controlname); - self->handle = 0; - } - PyObject_Del(self); -} - -static PyObject * -alsamixer_cardname(alsamixer_t *self, PyObject *args) { - if (!PyArg_ParseTuple(args,"")) return NULL; - return PyString_FromString(self->cardname); -} - -static PyObject * -alsamixer_mixer(alsamixer_t *self, PyObject *args) { - if (!PyArg_ParseTuple(args,"")) return NULL; - return PyString_FromString(self->controlname); -} - -static PyObject * -alsamixer_mixerid(alsamixer_t *self, PyObject *args) { - if (!PyArg_ParseTuple(args,"")) return NULL; - return PyInt_FromLong(self->controlid); -} - -static PyObject * -alsamixer_volumecap(alsamixer_t *self, PyObject *args) { - PyObject *result; - if (!PyArg_ParseTuple(args,"")) return NULL; - result = PyList_New(0); - if (self->volume_cap&MIXER_CAP_VOLUME) - PyList_Append(result,PyString_FromString("Volume")); - if (self->volume_cap&MIXER_CAP_VOLUME_JOINED) - PyList_Append(result,PyString_FromString("Joined Volume")); - if (self->volume_cap&MIXER_CAP_PVOLUME) - PyList_Append(result,PyString_FromString("Playback Volume")); - if (self->volume_cap&MIXER_CAP_PVOLUME_JOINED) - PyList_Append(result,PyString_FromString("Joined Playback Volume")); - if (self->volume_cap&MIXER_CAP_CVOLUME) - PyList_Append(result,PyString_FromString("Capture Volume")); - if (self->volume_cap&MIXER_CAP_CVOLUME_JOINED) - PyList_Append(result,PyString_FromString("Joined Capture Volume")); - - return result; -} -static PyObject * -alsamixer_switchcap(alsamixer_t *self, PyObject *args) { - PyObject *result; - if (!PyArg_ParseTuple(args,"")) return NULL; - result = PyList_New(0); - if (self->volume_cap&MIXER_CAP_SWITCH) - PyList_Append(result,PyString_FromString("Mute")); - if (self->volume_cap&MIXER_CAP_SWITCH_JOINED) - PyList_Append(result,PyString_FromString("Joined Mute")); - if (self->volume_cap&MIXER_CAP_PSWITCH) - PyList_Append(result,PyString_FromString("Playback Mute")); - if (self->volume_cap&MIXER_CAP_PSWITCH_JOINED) - PyList_Append(result,PyString_FromString("Joined Playback Mute")); - if (self->volume_cap&MIXER_CAP_CSWITCH) - PyList_Append(result,PyString_FromString("Capture Mute")); - if (self->volume_cap&MIXER_CAP_CSWITCH_JOINED) - PyList_Append(result,PyString_FromString("Joined Capture Mute")); - if (self->volume_cap&MIXER_CAP_CSWITCH_EXCLUSIVE) - PyList_Append(result,PyString_FromString("Capture Exclusive")); - return result; -} - -static int alsamixer_getpercentage(long min, long max, long value) { - /* Convert from number in range to percentage */ - int range = max - min; - int tmp; - - if (range == 0) return 0; - value -= min; - tmp = rint((double)value/(double)range * 100); - return tmp; - -} - -static long alsamixer_getphysvolume(long min, long max, int percentage) { - /* Convert from percentage to number in range */ - int range = max - min; - int tmp; - - if (range == 0) return 0; - tmp = rint((double)range * ((double)percentage*.01)) + min; - return tmp; -} - -static PyObject * -alsamixer_getvolume(alsamixer_t *self, PyObject *args) { - snd_mixer_elem_t *elem; - int direction; - int channel; - long ival; - 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 mixer"); - return NULL; - } - result = PyList_New(0); - for (channel = 0; channel <= SND_MIXER_SCHN_LAST; 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))); - } - 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))); - } - } - return result; -} - -static PyObject * -alsamixer_getmute(alsamixer_t *self, PyObject *args) { - snd_mixer_elem_t *elem; - int i; - int ival; - PyObject *result; - if (!PyArg_ParseTuple(args,"")) return NULL; - - elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid); - if (!snd_mixer_selem_has_playback_switch(elem)) { - PyErr_SetString(ALSAAudioError,"Mixer has no mute switch"); - return NULL; - } - result = PyList_New(0); - for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) { - if (snd_mixer_selem_has_playback_channel(elem, i)) { - snd_mixer_selem_get_playback_switch(elem, i, &ival); - PyList_Append(result,PyInt_FromLong(!ival)); - } - } - return result; -} - -static PyObject * -alsamixer_getrec(alsamixer_t *self, PyObject *args) { - snd_mixer_elem_t *elem; - int i; - int ival; - PyObject *result; - if (!PyArg_ParseTuple(args,"")) return NULL; - - elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid); - if (!snd_mixer_selem_has_capture_switch(elem)) { - PyErr_SetString(ALSAAudioError,"Mixer has no record switch"); - return NULL; - } - result = PyList_New(0); - for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) { - if (snd_mixer_selem_has_capture_channel(elem, i)) { - snd_mixer_selem_get_capture_switch(elem, i, &ival); - PyList_Append(result,PyInt_FromLong(!ival)); - } - } - return result; -} - -static PyObject * -alsamixer_setvolume(alsamixer_t *self, PyObject *args) { - snd_mixer_elem_t *elem; - int direction; - int i; - long volume; - int physvolume; - char *dirstr = 0; - int channel = MIXER_CHANNEL_ALL; - int done = 0; - - if (!PyArg_ParseTuple(args,"l|is",&volume,&channel,&dirstr)) return NULL; - if (volume < 0 || volume > 100) { - PyErr_SetString(ALSAAudioError,"Volume must be between 0 and 100"); - 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. Use 'playback' or 'capture'"); - return NULL; - } - for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) { - if (channel == -1 || channel == i) { - if (direction == 0 && snd_mixer_selem_has_playback_channel(elem, i)) { - physvolume = alsamixer_getphysvolume(self->pmin,self->pmax,volume); - snd_mixer_selem_set_playback_volume(elem, i, physvolume); - done++; - } - 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++; - } - } - } - if(!done) { - PyErr_SetString(ALSAAudioError,"No such channel"); - return NULL; - } - Py_INCREF(Py_None); - return Py_None; -} - -static PyObject * -alsamixer_setmute(alsamixer_t *self, PyObject *args) { - snd_mixer_elem_t *elem; - int i; - int mute = 0; - int done = 0; - int channel = MIXER_CHANNEL_ALL; - if (!PyArg_ParseTuple(args,"i|i",&mute,&channel)) return NULL; - - elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid); - if (!snd_mixer_selem_has_playback_switch(elem)) { - PyErr_SetString(ALSAAudioError,"Mixer has no mute switch"); - return NULL; - } - for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) { - if (channel == MIXER_CHANNEL_ALL || channel == i) { - if (snd_mixer_selem_has_playback_channel(elem, i)) { - snd_mixer_selem_set_playback_switch(elem, i, !mute); - done++; - } - } - } - if (!done) { - PyErr_SetString(ALSAAudioError,"Invalid channel number"); - return NULL; - } - Py_INCREF(Py_None); - return Py_None; -} - -static PyObject * -alsamixer_setrec(alsamixer_t *self, PyObject *args) { - snd_mixer_elem_t *elem; - int i; - int rec = 0; - int done = 0; - int channel = MIXER_CHANNEL_ALL; - if (!PyArg_ParseTuple(args,"i|i",&rec,&channel)) return NULL; - - elem = alsamixer_find_elem(self->handle,self->controlname,self->controlid); - if (!snd_mixer_selem_has_capture_switch(elem)) { - PyErr_SetString(ALSAAudioError,"Mixer has no record switch"); - return NULL; - } - for (i = 0; i <= SND_MIXER_SCHN_LAST; i++) { - if (channel == MIXER_CHANNEL_ALL || channel == i) { - if (snd_mixer_selem_has_capture_channel(elem, i)) { - snd_mixer_selem_set_playback_switch(elem, i, rec); - done++; - } - } - } - if (!done) { - PyErr_SetString(ALSAAudioError,"Invalid channel number"); - return NULL; - } - Py_INCREF(Py_None); - return Py_None; -} - -static PyMethodDef alsamixer_methods[] = { - {"cardname", (PyCFunction)alsamixer_cardname, METH_VARARGS}, - {"mixer", (PyCFunction)alsamixer_mixer, METH_VARARGS}, - {"mixerid", (PyCFunction)alsamixer_mixerid, METH_VARARGS}, - {"switchcap", (PyCFunction)alsamixer_switchcap, METH_VARARGS}, - {"volumecap", (PyCFunction)alsamixer_volumecap, METH_VARARGS}, - {"getvolume", (PyCFunction)alsamixer_getvolume, METH_VARARGS}, - {"getmute", (PyCFunction)alsamixer_getmute, METH_VARARGS}, - {"getrec", (PyCFunction)alsamixer_getrec, METH_VARARGS}, - {"setvolume", (PyCFunction)alsamixer_setvolume, METH_VARARGS}, - {"setmute", (PyCFunction)alsamixer_setmute, METH_VARARGS}, - {"setrec", (PyCFunction)alsamixer_setrec, METH_VARARGS}, - {NULL, NULL} -}; - - -static PyObject * -alsamixer_getattr(alsapcm_t *self, char *name) { - return Py_FindMethod(alsamixer_methods, (PyObject *)self, name); -} - -static PyTypeObject ALSAMixerType = { - PyObject_HEAD_INIT(&PyType_Type) - 0, /*ob_size*/ - "alsaaudio.mixer", /*tp_name*/ - sizeof(alsamixer_t), /*tp_basicsize*/ - 0, /*tp_itemsize*/ - /* methods */ - (destructor) alsamixer_dealloc, /*tp_dealloc*/ - 0, /*print*/ - (getattrfunc)alsamixer_getattr, /*tp_getattr*/ - 0, /*tp_setattr*/ - 0, /*tp_compare*/ - 0, /*tp_repr*/ - 0, /*tp_as_number*/ - 0, /*tp_as_sequence*/ - 0, /*tp_as_mapping*/ - 0, /*tp_hash*/ - 0, /*tp_call*/ - 0, /*tp_str*/ - 0, /*tp_getattro*/ - 0, /*tp_setattro*/ - 0, /*tp_as_buffer*/ - Py_TPFLAGS_DEFAULT, /*tp_flags*/ - "ALSA Mixer Control", /*tp_doc*/ -}; - - -/******************************************/ -/* Module initialization */ -/******************************************/ - -static PyMethodDef alsaaudio_methods[] = { - { "mixers", alsamixer_list, METH_VARARGS }, - { 0, 0 }, -}; - -#define _EXPORT_INT(mod, name, value) \ - if (PyModule_AddIntConstant(mod, name, (long) value) == -1) return; - -void initalsaaudio(void) { - PyObject *m; - ALSAPCMType.tp_new = alsapcm_new; - ALSAMixerType.tp_new = alsamixer_new; - m = Py_InitModule3("alsaaudio",alsaaudio_methods,alsaaudio_module_doc); - - ALSAAudioError = PyErr_NewException("alsaaudio.ALSAAudioError", NULL, NULL); - if (ALSAAudioError) { - /* Each call to PyModule_AddObject decrefs it; compensate: */ - - Py_INCREF(&ALSAPCMType); - PyModule_AddObject(m,"PCM",(PyObject *)&ALSAPCMType); - - Py_INCREF(&ALSAMixerType); - PyModule_AddObject(m,"Mixer",(PyObject *)&ALSAMixerType); - - Py_INCREF(ALSAAudioError); - PyModule_AddObject(m, "ALSAAudioError", ALSAAudioError); - } - - - _EXPORT_INT(m,"PCM_PLAYBACK",SND_PCM_STREAM_PLAYBACK); - _EXPORT_INT(m,"PCM_CAPTURE",SND_PCM_STREAM_CAPTURE); - - _EXPORT_INT(m,"PCM_NORMAL",0); - _EXPORT_INT(m,"PCM_NONBLOCK",SND_PCM_NONBLOCK); - _EXPORT_INT(m,"PCM_ASYNC",SND_PCM_ASYNC); - - /* PCM Formats */ - _EXPORT_INT(m,"PCM_FORMAT_S8",SND_PCM_FORMAT_S8); - _EXPORT_INT(m,"PCM_FORMAT_U8",SND_PCM_FORMAT_U8); - _EXPORT_INT(m,"PCM_FORMAT_S16_LE",SND_PCM_FORMAT_S16_LE); - _EXPORT_INT(m,"PCM_FORMAT_S16_BE",SND_PCM_FORMAT_S16_BE); - _EXPORT_INT(m,"PCM_FORMAT_U16_LE",SND_PCM_FORMAT_U16_LE); - _EXPORT_INT(m,"PCM_FORMAT_U16_BE",SND_PCM_FORMAT_U16_BE); - _EXPORT_INT(m,"PCM_FORMAT_S24_LE",SND_PCM_FORMAT_S24_LE); - _EXPORT_INT(m,"PCM_FORMAT_S24_BE",SND_PCM_FORMAT_S24_BE); - _EXPORT_INT(m,"PCM_FORMAT_U24_LE",SND_PCM_FORMAT_U24_LE); - _EXPORT_INT(m,"PCM_FORMAT_U24_BE",SND_PCM_FORMAT_U24_BE); - _EXPORT_INT(m,"PCM_FORMAT_S32_LE",SND_PCM_FORMAT_S32_LE); - _EXPORT_INT(m,"PCM_FORMAT_S32_BE",SND_PCM_FORMAT_S32_BE); - _EXPORT_INT(m,"PCM_FORMAT_U32_LE",SND_PCM_FORMAT_U32_LE); - _EXPORT_INT(m,"PCM_FORMAT_U32_BE",SND_PCM_FORMAT_U32_BE); - _EXPORT_INT(m,"PCM_FORMAT_FLOAT_LE",SND_PCM_FORMAT_FLOAT_LE); - _EXPORT_INT(m,"PCM_FORMAT_FLOAT_BE",SND_PCM_FORMAT_FLOAT_BE); - _EXPORT_INT(m,"PCM_FORMAT_FLOAT64_LE",SND_PCM_FORMAT_FLOAT64_LE); - _EXPORT_INT(m,"PCM_FORMAT_FLOAT64_BE",SND_PCM_FORMAT_FLOAT64_BE); - _EXPORT_INT(m,"PCM_FORMAT_MU_LAW",SND_PCM_FORMAT_MU_LAW); - _EXPORT_INT(m,"PCM_FORMAT_A_LAW",SND_PCM_FORMAT_A_LAW); - _EXPORT_INT(m,"PCM_FORMAT_IMA_ADPCM",SND_PCM_FORMAT_IMA_ADPCM); - _EXPORT_INT(m,"PCM_FORMAT_MPEG",SND_PCM_FORMAT_MPEG); - _EXPORT_INT(m,"PCM_FORMAT_GSM",SND_PCM_FORMAT_GSM); - - /* Mixer stuff */ - _EXPORT_INT(m,"MIXER_CHANNEL_ALL",MIXER_CHANNEL_ALL); - -} diff --git a/arch_src/pyalsaaudio-0.2/alsaaudio.o b/arch_src/pyalsaaudio-0.2/alsaaudio.o deleted file mode 100755 index ea1483d..0000000 --- a/arch_src/pyalsaaudio-0.2/alsaaudio.o +++ /dev/null Binary files differ diff --git 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diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/about.html.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/about.html.svn-base deleted file mode 100644 index c47ebca..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/about.html.svn-base +++ /dev/null @@ -1,110 +0,0 @@ - - - - - - - - - - - -About this document ... - - -
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-Front Matter -

- -

-This software is licensed under the PSF license - the same one used -by the majority of the python distribution. Basically you can use it -for anything you wish (even commercial purposes). There is no warranty -whatsoever. - -

- -

Abstract:

-
- -This package contains wrappers for accessing the ALSA api from Python. It -is currently fairly complete for PCM devices and Mixer access. MIDI sequencer -support is low on my priority list, but volunteers are welcome. - -

-If you find bugs in the wrappers please notify me on email. Please -don't send bug reports regarding ALSA specifically. There are several -bugs in this api, and those should be reported to the ALSA team - not -me. -

-

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- - - - diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.dat.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.dat.svn-base deleted file mode 100644 index 8efff3f..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.dat.svn-base +++ /dev/null @@ -1,27 +0,0 @@ -alsaaudio (extension module)###DEF0000000608 -mixers() (in module alsaaudio)###0000000609 -PCM (class in alsaaudio)###0000000610 -Mixer (class in alsaaudio)###0000000611 -ALSAAudioError###0000000612 -PCM (class in alsaaudio)###0000000622 -pcmtype() (PCM method)###0000000623 -pcmmode() (PCM method)###0000000624 -cardname() (PCM method)###0000000625 -setchannels() (PCM method)###0000000626 -setrate() (PCM method)###0000000627 -setformat() (PCM method)###0000000628 -setperiodsize() (PCM method)###0000000629 -read() (PCM method)###0000000630 -write() (PCM method)###0000000631 -Mixer (class in alsaaudio)###0000000633 -cardname() (Mixer method)###0000000634 -mixer() (Mixer method)###0000000635 -mixerid() (Mixer method)###0000000636 -switchcap() (Mixer method)###0000000637 -volumecap() (Mixer method)###0000000638 -getvolume() (Mixer method)###0000000639 -getmute() (Mixer method)###0000000640 -getrec() (Mixer method)###0000000641 -setvolume() (Mixer method)###0000000642 -setmute() (Mixer method)###0000000643 -setrec() (Mixer method)###0000000644 diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.gif.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.gif.svn-base deleted file mode 100644 index 32eecfb..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.gif.svn-base +++ /dev/null Binary files differ diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.html.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.html.svn-base deleted file mode 100644 index fc3e86f..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/index.html.svn-base +++ /dev/null @@ -1,110 +0,0 @@ - - - - - - - - - - -PyAlsaAudio - - -
- -
- - -
-
-

PyAlsaAudio

-

Casper Wilstrup

-

cwi@unispeed.com

-

-
-
- -

- -


-
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- - - - diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/internals.pl.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/internals.pl.svn-base deleted file mode 100644 index 4864f4b..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/internals.pl.svn-base +++ /dev/null @@ -1,34 +0,0 @@ -# LaTeX2HTML 2002-2 (1.70) -# Associate internals original text with physical files. - - -$key = q/module-alsaaudio/; -$ref_files{$key} = "$dir".q|node6.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-objects/; -$ref_files{$key} = "$dir".q|node8.html|; -$noresave{$key} = "$nosave"; - -$key = q/front/; -$ref_files{$key} = "$dir".q|node1.html|; -$noresave{$key} = "$nosave"; - -$key = q/contents/; -$ref_files{$key} = "$dir".q|node2.html|; -$noresave{$key} = "$nosave"; - -$key = q/mixer-objects/; -$ref_files{$key} = "$dir".q|node9.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-example/; -$ref_files{$key} = "$dir".q|node10.html|; -$noresave{$key} = "$nosave"; - -$key = q/about/; -$ref_files{$key} = "$dir".q|node11.html|; -$noresave{$key} = "$nosave"; - -1; - diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/intlabels.pl.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/intlabels.pl.svn-base deleted file mode 100644 index 972ec5a..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/intlabels.pl.svn-base +++ /dev/null @@ -1,30 +0,0 @@ -%internal_labels = (); -1; # hack in case there are no entries - -$internal_labels{"l2h-1"} = "/node6.html"; -$internal_labels{"l2h-2"} = "/node6.html"; -$internal_labels{"l2h-3"} = "/node6.html"; -$internal_labels{"l2h-4"} = "/node6.html"; -$internal_labels{"l2h-5"} = "/node6.html"; -$internal_labels{"l2h-6"} = "/node8.html"; -$internal_labels{"l2h-7"} = "/node8.html"; -$internal_labels{"l2h-8"} = "/node8.html"; -$internal_labels{"l2h-9"} = "/node8.html"; -$internal_labels{"l2h-10"} = "/node8.html"; -$internal_labels{"l2h-11"} = "/node8.html"; -$internal_labels{"l2h-12"} = "/node8.html"; -$internal_labels{"l2h-13"} = "/node8.html"; -$internal_labels{"l2h-14"} = "/node8.html"; -$internal_labels{"l2h-15"} = "/node8.html"; -$internal_labels{"l2h-16"} = "/node9.html"; -$internal_labels{"l2h-17"} = "/node9.html"; -$internal_labels{"l2h-18"} = "/node9.html"; -$internal_labels{"l2h-19"} = "/node9.html"; -$internal_labels{"l2h-20"} = "/node9.html"; -$internal_labels{"l2h-21"} = "/node9.html"; -$internal_labels{"l2h-22"} = "/node9.html"; -$internal_labels{"l2h-23"} = "/node9.html"; -$internal_labels{"l2h-24"} = "/node9.html"; -$internal_labels{"l2h-25"} = "/node9.html"; -$internal_labels{"l2h-26"} = "/node9.html"; -$internal_labels{"l2h-27"} = "/node9.html"; diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/labels.pl.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/labels.pl.svn-base deleted file mode 100644 index f3d0859..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/labels.pl.svn-base +++ /dev/null @@ -1,57 +0,0 @@ -# LaTeX2HTML 2002-2-1 (1.70) -# Associate labels original text with physical files. - - -$key = q/module-alsaaudio/; -$external_labels{$key} = "$URL/" . q|node6.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-objects/; -$external_labels{$key} = "$URL/" . q|node8.html|; -$noresave{$key} = "$nosave"; - -$key = q/front/; -$external_labels{$key} = "$URL/" . q|node1.html|; -$noresave{$key} = "$nosave"; - -$key = q/contents/; -$external_labels{$key} = "$URL/" . q|node2.html|; -$noresave{$key} = "$nosave"; - -$key = q/mixer-objects/; -$external_labels{$key} = "$URL/" . q|node9.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-example/; -$external_labels{$key} = "$URL/" . q|node10.html|; -$noresave{$key} = "$nosave"; - -$key = q/about/; -$external_labels{$key} = "$URL/" . q|node11.html|; -$noresave{$key} = "$nosave"; - -1; - - -# LaTeX2HTML 2002-2-1 (1.70) -# labels from external_latex_labels array. - - -$key = q/module-alsaaudio/; -$external_latex_labels{$key} = q|4|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-objects/; -$external_latex_labels{$key} = q|4.2|; -$noresave{$key} = "$nosave"; - -$key = q/mixer-objects/; -$external_latex_labels{$key} = q|4.3|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-example/; -$external_latex_labels{$key} = q|4.4|; -$noresave{$key} = "$nosave"; - -1; - diff --git a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/mixer-objects.html.svn-base b/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/mixer-objects.html.svn-base deleted file mode 100644 index 5a4d516..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/.svn/text-base/mixer-objects.html.svn-base +++ /dev/null @@ -1,327 +0,0 @@ - - - - - - - - - - - - - -4.3 Mixer Objects - - -
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-4.3 Mixer Objects -

- -

-Mixer objects provides access to the ALSA mixer API. - -

-

- -
class Mixer([control], [id], [cardname])
-
-control - specifies which control to manipulate using this mixer object. The list -of available controls can be found with the alsaaudio.mixers function. -The default value is 'Master' - other common controls include 'Master Mono', 'PCM', 'Line', etc. - -

-id - the id of the mixer control. Default is 0 - -

-cardname - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -

- -

-Mixer objects have the following methods: - -

-

- -
cardname()
-
-Return the name of the sound card used by this Mixer object -
- -

-

- -
mixer()
-
-Return the name of the specific mixer controlled by this object, For example 'Master' -or 'PCM' -
- -

-

- -
mixerid()
-
-Return the ID of the ALSA mixer controlled by this object. -
- -

-

- -
switchcap()
-
-Returns a list of the switches which are defined by this specific mixer. Possible values in -this list are: - -

-

- - - - - - - - - - - - - - - - - - - - - -
SwitchDescription
'Mute'This mixer can be muted
'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
'Joined Capture Mute'Mute sound capture for all channels at a time
'Capture Exclusive'Not quite sure what this is
- -

-To manipulate these swithes use the setrec or setmute methods -

- -

-

- -
volumecap()
-
-Returns a list of the volume control capabilities of this mixer. Possible values in -the list are: - -

-

- - - - - - - - - - - - - - - - - - - -
CapabilityDescription
'Volume'This mixer can control volume
'Joined Volume'This mixer can control volume for all channels at the same time
'Playback Volume'This mixer can manipulate the playback volume
'Joined Playback Volume'Manipulate playback volumne for all channels at the same time
'Capture Volume'Manipulate sound capture volume
'Joined Capture Volume'Manipulate sound capture volume for all channels at a time
- -

-

- -

-

- -
getvolume([direction])
-
-Returns a list with the current volume settings for each channel. The list elements -are integer percentages. - -

-The optional 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' - -

-

- -

-

- -
getmute()
-
-Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted. - -

-This method will fail if the mixer has no playback switch capabilities. -

- -

-

- -
getrec()
-
-Return a list indicating the current record mute setting for each channel. 0 means not recording, 1 -means not recording. - -

-This method will fail if the mixer has no capture switch capabilities. -

- -

-

- -
setvolume(volume,[channel],[direction])
-
-Change the current volume settings for this mixer. The volume argument controls -the new volume setting as an integer percentage. - -

-If the optional argument channel is present, the volume is set only for this channel. This -assumes that the mixer can control the volume for the channels independently. - -

-The optional direction argument can be either 'playback' or 'capture' is relevant if the mixer -has independent playback and capture volume capabilities, and controls which of the volumes -if changed. The default is 'playback' if the mixer has this capability, otherwise 'capture'. -

- -

-

- -
setmute(mute, [channel])
-
-Sets the mute flag to a new value. The mute argument is either 0 for not muted, or 1 for muted. - -

-The optional channel argument controls which channel is muted. The default is to set the mute flag -for all channels. - -

-This method will fail if the mixer has no playback mute capabilities -

- -

-

- -
setrec(capture,[channel])
-
-Sets the capture mute flag to a new value. The capture argument is either 0 for no capture, -or 1 for capture. - -

-The optional channel argument controls which channel is changed. The default is to set the capture flag -for all channels. - -

-This method will fail if the mixer has no capture switch capabilities -

- -

-A Note on the ALSA Mixer API - -

-The ALSA mixer API is extremely complicated - and hardly documented at all. alsaaudio implements -a much simplified way to access this API. In designing the API I've had to make some choices which -may limit what can and cannot be controlled through the API. However, If I had chosen to implement the -full API, I would have reexposed the horrible complexity/documentation ratio of the underlying API. -At least the alsaaudio API is easy to understand and use. - -

-If my design choises prevents you from doing something that the underlying API would have allowed, -please let me know, so I can incorporate these need into future versions. - -

-If the current state of affairs annoy you, the best you can do is to write a HOWTO on the API and -make this available on the net. Until somebody does this, the availability of ALSA mixer capable -devices will stay quite limited. - -

-Unfortunately, I'm not able to create such a HOWTO myself, since I only understand half of the API, -and that which I do understand has come from a painful trial and error process. - -

- -

-
-

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-4 alsaaudio -

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- -

- -

Availability: Linux.

- -

- -

- -

-The alsaaudio module defines functions and classes for using -ALSA. - -

-

- -
mixers([cardname])
-
-List the available mixers. The optional cardname specifies which -card should be queried (this is only relevant if you have more than one -sound card). Omit to use the default sound card -
- -

-

- -
class PCM([type], [mode], [cardname])
-
-This class is used to represent a PCM device (both playback and capture devices). -The arguments are: -
type - can be either PCM_CAPTURE or PCM_PLAYBACK (default). -
mode - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default). -
cardname - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -
- -

-

- -
class Mixer([control], [id], [cardname])
-
-This class is used to access a specific ALSA mixer. -The arguments are: -
control - Name of the chosen mixed (default is Master). -
id - id of mixer (default is 0) - More explaniation needed here -
cardname specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -
- -

-

exception ALSAAudioError
-
-Exception raised when an operation fails for a ALSA specific reason. -The exception argument is a string describing the reason of the -failure. -
- -

- -


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-1 What is ALSA -

- -

-The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI -functionality to the Linux operating system. - -

-Logically ALSA consists of these components: - -

    -
  • A set of kernel drivers. -
    -These drivers are responsible for handling the physical sound - hardware from within the Linux kernel, and have been the standard - sound implementation in Linux since kernel version 2.5 -
    • -
  • A kernel level API for manipulating the ALSA devices. -
    • -
  • A user-space C library for simplified access to the sound hardware - from userspace applications. This library is called libasound - and is required by all ALSA capable applications. -
    • -
- -

-More information about ALSA may be found on the project homepage -http://www.alsa-project.org -

- -

-
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-2 ALSA and Python -

- -

-The older Linux sound API (OSS) which is now deprecated is well supported -from the standard Python library, through the ossaudiodev module. No native -ALSA support exists in the standard library (yet). - -

-There are a few other ``ALSA for Python'' projects available, including at -least two different projects called pyAlsa. Neither of these seem to be under -active development at the time - and neither are very feature complete. - -

-I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module -included in the standard Python library, but that is probably a while of yet. - -

-PyAlsaAudio hass full support for sound capture, playback of sound, as well as -the ALSA Mixer API. - -

-MIDI support is not available, and since I don't own any MIDI hardware, it's -difficult for me to implement it. Volunteers to work on this would be greatly -appreciated - -

-
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-3 Installation -

- -

-Note: the wrappers link with the alsasound library (from the alsa-lib -package). Verify that this is installed by looking for /usr/lib/libasound.so -before building. Naturally you also need to use a kernel with proper ALSA -support. This is the default in Linux kernel 2.6 and later. If you are using -kernel version 2.4 you may need to install the ALSA patches yourself - although -most distributions ship with ALSA kernels. - -

-To install, execute the following: -

-$ python setup.py build
-
- -

-And then as root: -

-# python setup.py install
-
- -

- -

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-4.1 PCM Terminology and Concepts -

- -

-In order to use PCM devices it is useful to be familiar with some concepts and -terminology. - -

-

-
Sample
-
PCM audio, whether it is input or output, consists at the lowest level -of a number of single samples. A sample represents the sound in a single channel in -a brief interval. If more than one channel is in use, more than one sample is required -for each interval to describe the sound. Samples can be of many different sizes, ranging -from 8 bit to 64 bit presition. The specific format of each sample can also vary - they -can be big endian byte order, little endian byte order, or even floats. - -

-

-
Frame
-
A frame consists of exactly one sample per channel. If there is only one -channel (Mono sound) a frame is simply a single sample. If the sound is stereo, each frame -consists of two samples, etc. - -

-

-
Frame size
-
This is the size in bytes of each frame. This can vary a lot: if each sample is -8 bits, and we're handling mono sound, the frame size is one byte. Similarly in 6 channel audio with -64 bit floating point samples, the frame size is 48 bytes - -

-

-
Rate
-
PCM sound consists of a flow of sound frames. The sound rate controls how often -the current frame is replaced. For example, a rate of 8000 Hz means that a new frame is played -or captured 8000 times per second. - -

-

-
Data rate
-
This is the number of bytes, which must be recorded or provided per second -at a certain frame size and rate. - -

-8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s - -

-At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit (8 bytes) samples -has a data rate of 96000 * 6 * 8 = 4608 kb/s (almost 5 Mb sound data per second) - -

-

-
Period
-
When the hardware processes data this is done in chunks of frames. The time interval -between each processing (A/D or D/A conversion) is known as the period. The size of the period has -direct implication on the latency of the sound input or output. For low-latency the period size should -be very small, while low CPU resource usage would usually demand larger period sizes. With ALSA, the -CPU utilization is not impacted much by the period size, since the kernel layer buffers multiple -periods internally, so each period generates an interrupt and a memory copy, but userspace can be -slower and read or write multiple periods at the same time. - -

-

-
Period size
-
This is the size of each period in Hz. Not bytes, but Hz!. In alsaaudio -the period size is set directly, and it is therefore important to understand the significance of this -number. If the period size is configured to for example 32, each write should contain exactly 32 frames -of sound data, and each read will return either 32 frames of data or nothing at all. - -

-

-
- -

-Once you understand these concepts, you will be ready to actually utilize PCM API. Read on. - -

- -

-
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-4.4 ALSA Examples -

- -

-For now, the only examples available are the 'playbacktest.py' and the 'recordtest.py' programs included. -This will change in a future version. - -

- -

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-4.2 PCM Objects -

- -

-The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA -and many other places to handle playback and capture of sampled sound data. - -

-PCM objects in alsaaudio are used to do exactly that, either play sample based -sound or capture sound from some input source (perhaps a microphone). The PCM object -constructor takes the following arguments: - -

-

- -
class PCM([type], [mode], [cardname])
-
- -

-type - can be either PCM_CAPTURE or PCM_PLAYBACK (default). - -

-mode - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default). -In PCM_NONBLOCK mode, calls to read will return immediately independent of wether -there is any actual data to read. Similarly, write calls will return immediately -without actually writing anything to the playout buffer if the buffer is full. - -

-In the current version of alsaaudio PCM_ASYNC is useless, since it relies -on a callback procedure, which can't be specified from Python. - -

-cardname - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card - -

-This will construct a PCM object with default settings: - -

-Sample format: PCM_FORMAT_S16_LE -
-Rate: 8000 Hz -
-Channels: 2 -
-Period size: 32 frames -

- -

-PCM objects have the following methods: - -

-

- -
pcmtype()
-
-Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK. -
- -

-

- -
pcmmode()
-
-Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL -
- -

-

- -
cardname()
-
-Return the name of the sound card used by this PCM object. -
- -

-

- -
setchannels(nchannels)
-
-Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo, -and 6 = full 6 channel audio. Few sound cards support more than 2 channels -
- -

-

- -
setrate(rate)
-
-Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality), -and 96000 -
- -

-

- -
setformat()
-
-The sound format of the device. Sound format controls how the PCM device interpret data for playback, -and how data is encoded in captures. - -

-The following formats are provided by ALSA: -

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
FormatDescription
PCM_FORMAT_S8Signed 8 bit samples for each channel
PCM_FORMAT_U8Signed 8 bit samples for each channel
PCM_FORMAT_S16_LESigned 16 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S16_BESigned 16 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U16_LEUnsigned 16 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U16_BEUnsigned 16 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_S24_LESigned 24 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S24_BESigned 24 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U24_LEUnsigned 24 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U24_BEUnsigned 24 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_S32_LESigned 32 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S32_BESigned 32 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U32_LEUnsigned 32 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U32_BEUnsigned 32 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_FLOAT_LE32 bit samples encoded as float. (Little Endian byte order)
PCM_FORMAT_FLOAT_BE32 bit samples encoded as float (Big Endian byte order)
PCM_FORMAT_FLOAT64_LE64 bit samples encoded as float. (Little Endian byte order)
PCM_FORMAT_FLOAT64_BE64 bit samples encoded as float. (Big Endian byte order)
PCM_FORMAT_MU_LAWA logarithmic encoding (used by Sun .au files)
PCM_FORMAT_A_LAWAnother logarithmic encoding
PCM_FORMAT_IMA_ADPCMa 4:1 compressed format defined by the Interactive Multimedia Association
PCM_FORMAT_MPEGMPEG encoded audio?
PCM_FORMAT_GSM9600 constant rate encoding well suitet for speech
- -

-

- -

-

- -
setperiodsize(period)
-
-Sets the actual period size in frames. Each write should consist of exactly this number of frames, and -each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case -it may return nothing at all) -
- -

-

- -
read()
-
-In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a -tuple (length,data) where length is the size in bytes of the captured data, and data -is the captured sound frames as a string. The length of the returned data will be periodsize*framesize -bytes. - -

-In 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. -

- -

-

- -
write(data)
-
-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 PCM_NONBLOCK mode, this call will block if the kernel buffer is full, and -until enough sound has been played to allow the sound data to be buffered. The call always returns -the size of the data provided - -

-In PCM_NONBLOCK mode, the call will return immediately, with a return value of zero, if the buffer is -full. In this case, the data should be written at a later time. - -

-

- -

-A few hints on using PCM devices for playback - -

-The most common reason for problems with playback of PCM audio, is that the people don't properly understand -that writes to PCM devices must match exactly 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 written to the device, the write function will either block (PCM_NORMAL mode) or return zero -(PCM_NONBLOCK mode). - -

-If your program does nothing, but play sound, the easiest way is to put the device in PCM_NORMAL mode, and just -write as much data to the device as possible. This strategy can also be achieved by using a separate thread -with the sole task of playing out sound. - -

-In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few -periods by calling write a couple of times, and then use some timer method to write one period size of data to -the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer -doesn't expire exactly on time. - -

-Also note, that most timer API's that you can find for Python will cummulate time delays: If you set the timer -to expire after 1/10'th of a second, the actual timeout will happen slightly later, which will accumulate to -quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add -extra writes as nessecary. - -

- -

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-This software is licensed under the PSF license - the same one used -by the majority of the python distribution. Basically you can use it -for anything you wish (even commercial purposes). There is no warranty -whatsoever. - -

- -

Abstract:

-
- -This package contains wrappers for accessing the ALSA api from Python. It -is currently fairly complete for PCM devices and Mixer access. MIDI sequencer -support is low on my priority list, but volunteers are welcome. - -

-If you find bugs in the wrappers please notify me on email. Please -don't send bug reports regarding ALSA specifically. There are several -bugs in this api, and those should be reported to the ALSA team - not -me. -

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- - - - diff --git a/arch_src/pyalsaaudio-0.2/doc/index.dat b/arch_src/pyalsaaudio-0.2/doc/index.dat deleted file mode 100644 index 8efff3f..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/index.dat +++ /dev/null @@ -1,27 +0,0 @@ -alsaaudio (extension module)###DEF0000000608 -mixers() (in module alsaaudio)###0000000609 -PCM (class in alsaaudio)###0000000610 -Mixer (class in alsaaudio)###0000000611 -ALSAAudioError###0000000612 -PCM (class in alsaaudio)###0000000622 -pcmtype() (PCM method)###0000000623 -pcmmode() (PCM method)###0000000624 -cardname() (PCM method)###0000000625 -setchannels() (PCM method)###0000000626 -setrate() (PCM method)###0000000627 -setformat() (PCM method)###0000000628 -setperiodsize() (PCM method)###0000000629 -read() (PCM method)###0000000630 -write() (PCM method)###0000000631 -Mixer (class in alsaaudio)###0000000633 -cardname() (Mixer method)###0000000634 -mixer() (Mixer method)###0000000635 -mixerid() (Mixer method)###0000000636 -switchcap() (Mixer method)###0000000637 -volumecap() (Mixer method)###0000000638 -getvolume() (Mixer method)###0000000639 -getmute() (Mixer method)###0000000640 -getrec() (Mixer method)###0000000641 -setvolume() (Mixer method)###0000000642 -setmute() (Mixer method)###0000000643 -setrec() (Mixer method)###0000000644 diff --git a/arch_src/pyalsaaudio-0.2/doc/index.gif b/arch_src/pyalsaaudio-0.2/doc/index.gif deleted file mode 100644 index 32eecfb..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/index.gif +++ /dev/null Binary files differ diff --git a/arch_src/pyalsaaudio-0.2/doc/index.html b/arch_src/pyalsaaudio-0.2/doc/index.html deleted file mode 100644 index fc3e86f..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/index.html +++ /dev/null @@ -1,110 +0,0 @@ - - - - - - - - - - -PyAlsaAudio - - -
- -
- - -
-
-

PyAlsaAudio

-

Casper Wilstrup

-

cwi@unispeed.com

-

-
-
- -

- -


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- - - - diff --git a/arch_src/pyalsaaudio-0.2/doc/index.png b/arch_src/pyalsaaudio-0.2/doc/index.png deleted file mode 100644 index cd918af..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/index.png +++ /dev/null Binary files differ diff --git a/arch_src/pyalsaaudio-0.2/doc/internals.pl b/arch_src/pyalsaaudio-0.2/doc/internals.pl deleted file mode 100644 index 4864f4b..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/internals.pl +++ /dev/null @@ -1,34 +0,0 @@ -# LaTeX2HTML 2002-2 (1.70) -# Associate internals original text with physical files. - - -$key = q/module-alsaaudio/; -$ref_files{$key} = "$dir".q|node6.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-objects/; -$ref_files{$key} = "$dir".q|node8.html|; -$noresave{$key} = "$nosave"; - -$key = q/front/; -$ref_files{$key} = "$dir".q|node1.html|; -$noresave{$key} = "$nosave"; - -$key = q/contents/; -$ref_files{$key} = "$dir".q|node2.html|; -$noresave{$key} = "$nosave"; - -$key = q/mixer-objects/; -$ref_files{$key} = "$dir".q|node9.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-example/; -$ref_files{$key} = "$dir".q|node10.html|; -$noresave{$key} = "$nosave"; - -$key = q/about/; -$ref_files{$key} = "$dir".q|node11.html|; -$noresave{$key} = "$nosave"; - -1; - diff --git a/arch_src/pyalsaaudio-0.2/doc/intlabels.pl b/arch_src/pyalsaaudio-0.2/doc/intlabels.pl deleted file mode 100644 index 972ec5a..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/intlabels.pl +++ /dev/null @@ -1,30 +0,0 @@ -%internal_labels = (); -1; # hack in case there are no entries - -$internal_labels{"l2h-1"} = "/node6.html"; -$internal_labels{"l2h-2"} = "/node6.html"; -$internal_labels{"l2h-3"} = "/node6.html"; -$internal_labels{"l2h-4"} = "/node6.html"; -$internal_labels{"l2h-5"} = "/node6.html"; -$internal_labels{"l2h-6"} = "/node8.html"; -$internal_labels{"l2h-7"} = "/node8.html"; -$internal_labels{"l2h-8"} = "/node8.html"; -$internal_labels{"l2h-9"} = "/node8.html"; -$internal_labels{"l2h-10"} = "/node8.html"; -$internal_labels{"l2h-11"} = "/node8.html"; -$internal_labels{"l2h-12"} = "/node8.html"; -$internal_labels{"l2h-13"} = "/node8.html"; -$internal_labels{"l2h-14"} = "/node8.html"; -$internal_labels{"l2h-15"} = "/node8.html"; -$internal_labels{"l2h-16"} = "/node9.html"; -$internal_labels{"l2h-17"} = "/node9.html"; -$internal_labels{"l2h-18"} = "/node9.html"; -$internal_labels{"l2h-19"} = "/node9.html"; -$internal_labels{"l2h-20"} = "/node9.html"; -$internal_labels{"l2h-21"} = "/node9.html"; -$internal_labels{"l2h-22"} = "/node9.html"; -$internal_labels{"l2h-23"} = "/node9.html"; -$internal_labels{"l2h-24"} = "/node9.html"; -$internal_labels{"l2h-25"} = "/node9.html"; -$internal_labels{"l2h-26"} = "/node9.html"; -$internal_labels{"l2h-27"} = "/node9.html"; diff --git a/arch_src/pyalsaaudio-0.2/doc/labels.pl b/arch_src/pyalsaaudio-0.2/doc/labels.pl deleted file mode 100644 index f3d0859..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/labels.pl +++ /dev/null @@ -1,57 +0,0 @@ -# LaTeX2HTML 2002-2-1 (1.70) -# Associate labels original text with physical files. - - -$key = q/module-alsaaudio/; -$external_labels{$key} = "$URL/" . q|node6.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-objects/; -$external_labels{$key} = "$URL/" . q|node8.html|; -$noresave{$key} = "$nosave"; - -$key = q/front/; -$external_labels{$key} = "$URL/" . q|node1.html|; -$noresave{$key} = "$nosave"; - -$key = q/contents/; -$external_labels{$key} = "$URL/" . q|node2.html|; -$noresave{$key} = "$nosave"; - -$key = q/mixer-objects/; -$external_labels{$key} = "$URL/" . q|node9.html|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-example/; -$external_labels{$key} = "$URL/" . q|node10.html|; -$noresave{$key} = "$nosave"; - -$key = q/about/; -$external_labels{$key} = "$URL/" . q|node11.html|; -$noresave{$key} = "$nosave"; - -1; - - -# LaTeX2HTML 2002-2-1 (1.70) -# labels from external_latex_labels array. - - -$key = q/module-alsaaudio/; -$external_latex_labels{$key} = q|4|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-objects/; -$external_latex_labels{$key} = q|4.2|; -$noresave{$key} = "$nosave"; - -$key = q/mixer-objects/; -$external_latex_labels{$key} = q|4.3|; -$noresave{$key} = "$nosave"; - -$key = q/pcm-example/; -$external_latex_labels{$key} = q|4.4|; -$noresave{$key} = "$nosave"; - -1; - diff --git a/arch_src/pyalsaaudio-0.2/doc/mixer-objects.html b/arch_src/pyalsaaudio-0.2/doc/mixer-objects.html deleted file mode 100644 index 5a4d516..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/mixer-objects.html +++ /dev/null @@ -1,327 +0,0 @@ - - - - - - - - - - - - - -4.3 Mixer Objects - - -
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-4.3 Mixer Objects -

- -

-Mixer objects provides access to the ALSA mixer API. - -

-

- -
class Mixer([control], [id], [cardname])
-
-control - specifies which control to manipulate using this mixer object. The list -of available controls can be found with the alsaaudio.mixers function. -The default value is 'Master' - other common controls include 'Master Mono', 'PCM', 'Line', etc. - -

-id - the id of the mixer control. Default is 0 - -

-cardname - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -

- -

-Mixer objects have the following methods: - -

-

- -
cardname()
-
-Return the name of the sound card used by this Mixer object -
- -

-

- -
mixer()
-
-Return the name of the specific mixer controlled by this object, For example 'Master' -or 'PCM' -
- -

-

- -
mixerid()
-
-Return the ID of the ALSA mixer controlled by this object. -
- -

-

- -
switchcap()
-
-Returns a list of the switches which are defined by this specific mixer. Possible values in -this list are: - -

-

- - - - - - - - - - - - - - - - - - - - - -
SwitchDescription
'Mute'This mixer can be muted
'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
'Joined Capture Mute'Mute sound capture for all channels at a time
'Capture Exclusive'Not quite sure what this is
- -

-To manipulate these swithes use the setrec or setmute methods -

- -

-

- -
volumecap()
-
-Returns a list of the volume control capabilities of this mixer. Possible values in -the list are: - -

-

- - - - - - - - - - - - - - - - - - - -
CapabilityDescription
'Volume'This mixer can control volume
'Joined Volume'This mixer can control volume for all channels at the same time
'Playback Volume'This mixer can manipulate the playback volume
'Joined Playback Volume'Manipulate playback volumne for all channels at the same time
'Capture Volume'Manipulate sound capture volume
'Joined Capture Volume'Manipulate sound capture volume for all channels at a time
- -

-

- -

-

- -
getvolume([direction])
-
-Returns a list with the current volume settings for each channel. The list elements -are integer percentages. - -

-The optional 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' - -

-

- -

-

- -
getmute()
-
-Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted. - -

-This method will fail if the mixer has no playback switch capabilities. -

- -

-

- -
getrec()
-
-Return a list indicating the current record mute setting for each channel. 0 means not recording, 1 -means not recording. - -

-This method will fail if the mixer has no capture switch capabilities. -

- -

-

- -
setvolume(volume,[channel],[direction])
-
-Change the current volume settings for this mixer. The volume argument controls -the new volume setting as an integer percentage. - -

-If the optional argument channel is present, the volume is set only for this channel. This -assumes that the mixer can control the volume for the channels independently. - -

-The optional direction argument can be either 'playback' or 'capture' is relevant if the mixer -has independent playback and capture volume capabilities, and controls which of the volumes -if changed. The default is 'playback' if the mixer has this capability, otherwise 'capture'. -

- -

-

- -
setmute(mute, [channel])
-
-Sets the mute flag to a new value. The mute argument is either 0 for not muted, or 1 for muted. - -

-The optional channel argument controls which channel is muted. The default is to set the mute flag -for all channels. - -

-This method will fail if the mixer has no playback mute capabilities -

- -

-

- -
setrec(capture,[channel])
-
-Sets the capture mute flag to a new value. The capture argument is either 0 for no capture, -or 1 for capture. - -

-The optional channel argument controls which channel is changed. The default is to set the capture flag -for all channels. - -

-This method will fail if the mixer has no capture switch capabilities -

- -

-A Note on the ALSA Mixer API - -

-The ALSA mixer API is extremely complicated - and hardly documented at all. alsaaudio implements -a much simplified way to access this API. In designing the API I've had to make some choices which -may limit what can and cannot be controlled through the API. However, If I had chosen to implement the -full API, I would have reexposed the horrible complexity/documentation ratio of the underlying API. -At least the alsaaudio API is easy to understand and use. - -

-If my design choises prevents you from doing something that the underlying API would have allowed, -please let me know, so I can incorporate these need into future versions. - -

-If the current state of affairs annoy you, the best you can do is to write a HOWTO on the API and -make this available on the net. Until somebody does this, the availability of ALSA mixer capable -devices will stay quite limited. - -

-Unfortunately, I'm not able to create such a HOWTO myself, since I only understand half of the API, -and that which I do understand has come from a painful trial and error process. - -

- -

-
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-4 alsaaudio -

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- -

- -

Availability: Linux.

- -

- -

- -

-The alsaaudio module defines functions and classes for using -ALSA. - -

-

- -
mixers([cardname])
-
-List the available mixers. The optional cardname specifies which -card should be queried (this is only relevant if you have more than one -sound card). Omit to use the default sound card -
- -

-

- -
class PCM([type], [mode], [cardname])
-
-This class is used to represent a PCM device (both playback and capture devices). -The arguments are: -
type - can be either PCM_CAPTURE or PCM_PLAYBACK (default). -
mode - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default). -
cardname - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -
- -

-

- -
class Mixer([control], [id], [cardname])
-
-This class is used to access a specific ALSA mixer. -The arguments are: -
control - Name of the chosen mixed (default is Master). -
id - id of mixer (default is 0) - More explaniation needed here -
cardname specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -
- -

-

exception ALSAAudioError
-
-Exception raised when an operation fails for a ALSA specific reason. -The exception argument is a string describing the reason of the -failure. -
- -

- -


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-1 What is ALSA -

- -

-The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI -functionality to the Linux operating system. - -

-Logically ALSA consists of these components: - -

    -
  • A set of kernel drivers. -
    -These drivers are responsible for handling the physical sound - hardware from within the Linux kernel, and have been the standard - sound implementation in Linux since kernel version 2.5 -
    • -
  • A kernel level API for manipulating the ALSA devices. -
    • -
  • A user-space C library for simplified access to the sound hardware - from userspace applications. This library is called libasound - and is required by all ALSA capable applications. -
    • -
- -

-More information about ALSA may be found on the project homepage -http://www.alsa-project.org -

- -

-
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-2 ALSA and Python -

- -

-The older Linux sound API (OSS) which is now deprecated is well supported -from the standard Python library, through the ossaudiodev module. No native -ALSA support exists in the standard library (yet). - -

-There are a few other ``ALSA for Python'' projects available, including at -least two different projects called pyAlsa. Neither of these seem to be under -active development at the time - and neither are very feature complete. - -

-I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module -included in the standard Python library, but that is probably a while of yet. - -

-PyAlsaAudio hass full support for sound capture, playback of sound, as well as -the ALSA Mixer API. - -

-MIDI support is not available, and since I don't own any MIDI hardware, it's -difficult for me to implement it. Volunteers to work on this would be greatly -appreciated - -

-
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-3 Installation -

- -

-Note: the wrappers link with the alsasound library (from the alsa-lib -package). Verify that this is installed by looking for /usr/lib/libasound.so -before building. Naturally you also need to use a kernel with proper ALSA -support. This is the default in Linux kernel 2.6 and later. If you are using -kernel version 2.4 you may need to install the ALSA patches yourself - although -most distributions ship with ALSA kernels. - -

-To install, execute the following: -

-$ python setup.py build
-
- -

-And then as root: -

-# python setup.py install
-
- -

- -

-
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-4.1 PCM Terminology and Concepts -

- -

-In order to use PCM devices it is useful to be familiar with some concepts and -terminology. - -

-

-
Sample
-
PCM audio, whether it is input or output, consists at the lowest level -of a number of single samples. A sample represents the sound in a single channel in -a brief interval. If more than one channel is in use, more than one sample is required -for each interval to describe the sound. Samples can be of many different sizes, ranging -from 8 bit to 64 bit presition. The specific format of each sample can also vary - they -can be big endian byte order, little endian byte order, or even floats. - -

-

-
Frame
-
A frame consists of exactly one sample per channel. If there is only one -channel (Mono sound) a frame is simply a single sample. If the sound is stereo, each frame -consists of two samples, etc. - -

-

-
Frame size
-
This is the size in bytes of each frame. This can vary a lot: if each sample is -8 bits, and we're handling mono sound, the frame size is one byte. Similarly in 6 channel audio with -64 bit floating point samples, the frame size is 48 bytes - -

-

-
Rate
-
PCM sound consists of a flow of sound frames. The sound rate controls how often -the current frame is replaced. For example, a rate of 8000 Hz means that a new frame is played -or captured 8000 times per second. - -

-

-
Data rate
-
This is the number of bytes, which must be recorded or provided per second -at a certain frame size and rate. - -

-8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s - -

-At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit (8 bytes) samples -has a data rate of 96000 * 6 * 8 = 4608 kb/s (almost 5 Mb sound data per second) - -

-

-
Period
-
When the hardware processes data this is done in chunks of frames. The time interval -between each processing (A/D or D/A conversion) is known as the period. The size of the period has -direct implication on the latency of the sound input or output. For low-latency the period size should -be very small, while low CPU resource usage would usually demand larger period sizes. With ALSA, the -CPU utilization is not impacted much by the period size, since the kernel layer buffers multiple -periods internally, so each period generates an interrupt and a memory copy, but userspace can be -slower and read or write multiple periods at the same time. - -

-

-
Period size
-
This is the size of each period in Hz. Not bytes, but Hz!. In alsaaudio -the period size is set directly, and it is therefore important to understand the significance of this -number. If the period size is configured to for example 32, each write should contain exactly 32 frames -of sound data, and each read will return either 32 frames of data or nothing at all. - -

-

-
- -

-Once you understand these concepts, you will be ready to actually utilize PCM API. Read on. - -

- -

-
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-4.4 ALSA Examples -

- -

-For now, the only examples available are the 'playbacktest.py' and the 'recordtest.py' programs included. -This will change in a future version. - -

- -

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-4.2 PCM Objects -

- -

-The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA -and many other places to handle playback and capture of sampled sound data. - -

-PCM objects in alsaaudio are used to do exactly that, either play sample based -sound or capture sound from some input source (perhaps a microphone). The PCM object -constructor takes the following arguments: - -

-

- -
class PCM([type], [mode], [cardname])
-
- -

-type - can be either PCM_CAPTURE or PCM_PLAYBACK (default). - -

-mode - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default). -In PCM_NONBLOCK mode, calls to read will return immediately independent of wether -there is any actual data to read. Similarly, write calls will return immediately -without actually writing anything to the playout buffer if the buffer is full. - -

-In the current version of alsaaudio PCM_ASYNC is useless, since it relies -on a callback procedure, which can't be specified from Python. - -

-cardname - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card - -

-This will construct a PCM object with default settings: - -

-Sample format: PCM_FORMAT_S16_LE -
-Rate: 8000 Hz -
-Channels: 2 -
-Period size: 32 frames -

- -

-PCM objects have the following methods: - -

-

- -
pcmtype()
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-Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK. -
- -

-

- -
pcmmode()
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-Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL -
- -

-

- -
cardname()
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-Return the name of the sound card used by this PCM object. -
- -

-

- -
setchannels(nchannels)
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-Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo, -and 6 = full 6 channel audio. Few sound cards support more than 2 channels -
- -

-

- -
setrate(rate)
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-Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality), -and 96000 -
- -

-

- -
setformat()
-
-The sound format of the device. Sound format controls how the PCM device interpret data for playback, -and how data is encoded in captures. - -

-The following formats are provided by ALSA: -

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
FormatDescription
PCM_FORMAT_S8Signed 8 bit samples for each channel
PCM_FORMAT_U8Signed 8 bit samples for each channel
PCM_FORMAT_S16_LESigned 16 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S16_BESigned 16 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U16_LEUnsigned 16 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U16_BEUnsigned 16 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_S24_LESigned 24 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S24_BESigned 24 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U24_LEUnsigned 24 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U24_BEUnsigned 24 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_S32_LESigned 32 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S32_BESigned 32 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U32_LEUnsigned 32 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U32_BEUnsigned 32 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_FLOAT_LE32 bit samples encoded as float. (Little Endian byte order)
PCM_FORMAT_FLOAT_BE32 bit samples encoded as float (Big Endian byte order)
PCM_FORMAT_FLOAT64_LE64 bit samples encoded as float. (Little Endian byte order)
PCM_FORMAT_FLOAT64_BE64 bit samples encoded as float. (Big Endian byte order)
PCM_FORMAT_MU_LAWA logarithmic encoding (used by Sun .au files)
PCM_FORMAT_A_LAWAnother logarithmic encoding
PCM_FORMAT_IMA_ADPCMa 4:1 compressed format defined by the Interactive Multimedia Association
PCM_FORMAT_MPEGMPEG encoded audio?
PCM_FORMAT_GSM9600 constant rate encoding well suitet for speech
- -

-

- -

-

- -
setperiodsize(period)
-
-Sets the actual period size in frames. Each write should consist of exactly this number of frames, and -each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case -it may return nothing at all) -
- -

-

- -
read()
-
-In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a -tuple (length,data) where length is the size in bytes of the captured data, and data -is the captured sound frames as a string. The length of the returned data will be periodsize*framesize -bytes. - -

-In 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. -

- -

-

- -
write(data)
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-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 PCM_NONBLOCK mode, this call will block if the kernel buffer is full, and -until enough sound has been played to allow the sound data to be buffered. The call always returns -the size of the data provided - -

-In PCM_NONBLOCK mode, the call will return immediately, with a return value of zero, if the buffer is -full. In this case, the data should be written at a later time. - -

-

- -

-A few hints on using PCM devices for playback - -

-The most common reason for problems with playback of PCM audio, is that the people don't properly understand -that writes to PCM devices must match exactly 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 written to the device, the write function will either block (PCM_NORMAL mode) or return zero -(PCM_NONBLOCK mode). - -

-If your program does nothing, but play sound, the easiest way is to put the device in PCM_NORMAL mode, and just -write as much data to the device as possible. This strategy can also be achieved by using a separate thread -with the sole task of playing out sound. - -

-In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few -periods by calling write a couple of times, and then use some timer method to write one period size of data to -the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer -doesn't expire exactly on time. - -

-Also note, that most timer API's that you can find for Python will cummulate time delays: If you set the timer -to expire after 1/10'th of a second, the actual timeout will happen slightly later, which will accumulate to -quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add -extra writes as nessecary. - -

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Set the source path below -# -# You also need a working latex installation, and the latex2html -# tool installed. -PYTHONSOURCE = /usr/src/Python-2.4.1/ - -# Shouldn't need to change anything below here! - -MKHOWTO = $(PYTHONSOURCE)/Doc/tools/mkhowto - -all: - $(MKHOWTO) --dir .. --html pyalsaaudio.tex - -text: - $(MKHOWTO) --dir .. --text pyalsaaudio.tex - - diff --git a/arch_src/pyalsaaudio-0.2/doc/src/.svn/text-base/libalsaaudio.tex.svn-base b/arch_src/pyalsaaudio-0.2/doc/src/.svn/text-base/libalsaaudio.tex.svn-base deleted file mode 100644 index c50ffe5..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/.svn/text-base/libalsaaudio.tex.svn-base +++ /dev/null @@ -1,397 +0,0 @@ -\section{\module{alsaaudio}} - -%\declaremodule{builtin}{alsaaudio} % standard library, in C -\declaremodule{extension}{alsaaudio} % not standard, in C - -\platform{Linux} - -\moduleauthor{Casper Wilstrup}{cwi@unispeed.com} % Author of the module code; - - -\modulesynopsis{ALSA sound support} - - -The \module{alsaaudio} module defines functions and classes for using -ALSA. - -% ---- 3.1. ---- -% For each function, use a ``funcdesc'' block. This has exactly two -% parameters (each parameters is contained in a set of curly braces): -% the first parameter is the function name (this automatically -% generates an index entry); the second parameter is the function's -% argument list. If there are no arguments, use an empty pair of -% curly braces. If there is more than one argument, separate the -% arguments with backslash-comma. Optional parts of the parameter -% list are contained in \optional{...} (this generates a set of square -% brackets around its parameter). Arguments are automatically set in -% italics in the parameter list. Each argument should be mentioned at -% least once in the description; each usage (even inside \code{...}) -% should be enclosed in \var{...}. - -\begin{funcdesc}{mixers}{\optional{cardname}} -List the available mixers. The optional \var{cardname} specifies which -card should be queried (this is only relevant if you have more than one -sound card). Omit to use the default sound card -\end{funcdesc} - -\begin{classdesc}{PCM}{\optional{type}, \optional{mode}, \optional{cardname}} -This class is used to represent a PCM device (both playback and capture devices). -The arguments are: \\ -\var{type} - can be either PCM_CAPTURE or PCM_PLAYBACK (default). \\ -\var{mode} - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).\\ -\var{cardname} - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -\end{classdesc} - -\begin{classdesc}{Mixer}{\optional{control}, \optional{id}, \optional{cardname}} -This class is used to access a specific ALSA mixer. -The arguments are: \\ -\var{control} - Name of the chosen mixed (default is Master). \\ -\var{id} - id of mixer (default is 0) -- More explaniation needed here\\ -\var{cardname} specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -\end{classdesc} - - -\begin{excdesc}{ALSAAudioError} -Exception raised when an operation fails for a ALSA specific reason. -The exception argument is a string describing the reason of the -failure. -\end{excdesc} - -\subsection{PCM Terminology and Concepts} - -In order to use PCM devices it is useful to be familiar with some concepts and -terminology. - -\begin{description} -\item[Sample] PCM audio, whether it is input or output, consists at the lowest level -of a number of single samples. A sample represents the sound in a single channel in -a brief interval. If more than one channel is in use, more than one sample is required -for each interval to describe the sound. Samples can be of many different sizes, ranging -from 8 bit to 64 bit presition. The specific format of each sample can also vary - they -can be big endian byte order, little endian byte order, or even floats. - -\item[Frame] A frame consists of exactly one sample per channel. If there is only one -channel (Mono sound) a frame is simply a single sample. If the sound is stereo, each frame -consists of two samples, etc. - -\item[Frame size] This is the size in bytes of each frame. This can vary a lot: if each sample is -8 bits, and we're handling mono sound, the frame size is one byte. Similarly in 6 channel audio with -64 bit floating point samples, the frame size is 48 bytes - -\item[Rate] PCM sound consists of a flow of sound frames. The sound rate controls how often -the current frame is replaced. For example, a rate of 8000 Hz means that a new frame is played -or captured 8000 times per second. - -\item[Data rate] This is the number of bytes, which must be recorded or provided per second -at a certain frame size and rate. - -8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s - -At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit (8 bytes) samples -has a data rate of 96000 * 6 * 8 = 4608 kb/s (almost 5 Mb sound data per second) - -\item[Period] When the hardware processes data this is done in chunks of frames. The time interval -between each processing (A/D or D/A conversion) is known as the period. The size of the period has -direct implication on the latency of the sound input or output. For low-latency the period size should -be very small, while low CPU resource usage would usually demand larger period sizes. With ALSA, the -CPU utilization is not impacted much by the period size, since the kernel layer buffers multiple -periods internally, so each period generates an interrupt and a memory copy, but userspace can be -slower and read or write multiple periods at the same time. - -\item[Period size] This is the size of each period in Hz. \emph{Not bytes, but Hz!.} In \module{alsaaudio} -the period size is set directly, and it is therefore important to understand the significance of this -number. If the period size is configured to for example 32, each write should contain exactly 32 frames -of sound data, and each read will return either 32 frames of data or nothing at all. - -\end{description} - -Once you understand these concepts, you will be ready to actually utilize PCM API. Read on. - -\subsection{PCM Objects} -\label{pcm-objects} - -The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA -and many other places to handle playback and capture of sampled sound data. - -PCM objects in \module{alsaaudio} are used to do exactly that, either play sample based -sound or capture sound from some input source (perhaps a microphone). The PCM object -constructor takes the following arguments: - -\begin{classdesc}{PCM}{\optional{type}, \optional{mode}, \optional{cardname}} - -\var{type} - can be either PCM_CAPTURE or PCM_PLAYBACK (default). - -\var{mode} - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default). -In PCM_NONBLOCK mode, calls to read will return immediately independent of wether -there is any actual data to read. Similarly, write calls will return immediately -without actually writing anything to the playout buffer if the buffer is full. - -In the current version of \module{alsaaudio} PCM_ASYNC is useless, since it relies -on a callback procedure, which can't be specified from Python. - -\var{cardname} - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card - -This will construct a PCM object with default settings: - -Sample format: PCM_FORMAT_S16_LE \\ -Rate: 8000 Hz \\ -Channels: 2 \\ -Period size: 32 frames \\ -\end{classdesc} - -PCM objects have the following methods: - -\begin{methoddesc}[PCM]{pcmtype}{} -Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK. -\end{methoddesc} - -\begin{methoddesc}[PCM]{pcmmode}{} -Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL -\end{methoddesc} - -\begin{methoddesc}[PCM]{cardname}{} -Return the name of the sound card used by this PCM object. -\end{methoddesc} - -\begin{methoddesc}[PCM]{setchannels}{nchannels} -Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo, -and 6 = full 6 channel audio. Few sound cards support more than 2 channels -\end{methoddesc} - -\begin{methoddesc}[PCM]{setrate}{rate} -Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality), -and 96000 -\end{methoddesc} - -\begin{methoddesc}[PCM]{setformat}{} -The sound format of the device. Sound format controls how the PCM device interpret data for playback, -and how data is encoded in captures. - -The following formats are provided by ALSA: -\begin{tableii}{l|l}{Formats}{Format}{Description} - \lineii{PCM_FORMAT_S8}{Signed 8 bit samples for each channel} - \lineii{PCM_FORMAT_U8}{Signed 8 bit samples for each channel} - \lineii{PCM_FORMAT_S16_LE}{Signed 16 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_S16_BE}{Signed 16 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_U16_LE}{Unsigned 16 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_U16_BE}{Unsigned 16 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_S24_LE}{Signed 24 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_S24_BE}{Signed 24 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_U24_LE}{Unsigned 24 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_U24_BE}{Unsigned 24 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_S32_LE}{Signed 32 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_S32_BE}{Signed 32 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_U32_LE}{Unsigned 32 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_U32_BE}{Unsigned 32 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_FLOAT_LE}{32 bit samples encoded as float. (Little Endian byte order)} - \lineii{PCM_FORMAT_FLOAT_BE}{32 bit samples encoded as float (Big Endian byte order)} - \lineii{PCM_FORMAT_FLOAT64_LE}{64 bit samples encoded as float. (Little Endian byte order)} - \lineii{PCM_FORMAT_FLOAT64_BE}{64 bit samples encoded as float. (Big Endian byte order)} - \lineii{PCM_FORMAT_MU_LAW}{A logarithmic encoding (used by Sun .au files)} - \lineii{PCM_FORMAT_A_LAW}{Another logarithmic encoding} - \lineii{PCM_FORMAT_IMA_ADPCM}{a 4:1 compressed format defined by the Interactive Multimedia Association} - \lineii{PCM_FORMAT_MPEG}{MPEG encoded audio?} - \lineii{PCM_FORMAT_GSM}{9600 constant rate encoding well suitet for speech} -\end{tableii} - -\end{methoddesc} - -\begin{methoddesc}[PCM]{setperiodsize}{period} -Sets the actual period size in frames. Each write should consist of exactly this number of frames, and -each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case -it may return nothing at all) -\end{methoddesc} - -\begin{methoddesc}[PCM]{read}{} -In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a -tuple (length,data) where \emph{length} is the size in bytes of the captured data, and \emph{data} -is the captured sound frames as a string. The length of the returned data will be periodsize*framesize -bytes. - -In PCM_NONBLOCK mode, the call will not block, but will return \code{(0,'')} if no new period -has become available since the last call to read. -\end{methoddesc} - -\begin{methoddesc}[PCM]{write}{data} -Writes (plays) the sound in data. The length of data \emph{must} be a multiple of the frame size, and -\emph{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 PCM_NONBLOCK mode, this call will block if the kernel buffer is full, and -until enough sound has been played to allow the sound data to be buffered. The call always returns -the size of the data provided - -In PCM_NONBLOCK mode, the call will return immediately, with a return value of zero, if the buffer is -full. In this case, the data should be written at a later time. - -\end{methoddesc} - -\strong{A few hints on using PCM devices for playback} - -The most common reason for problems with playback of PCM audio, is that the people don't properly understand -that writes to PCM devices must match \emph{exactly} 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 written to the device, the write function will either block (PCM_NORMAL mode) or return zero -(PCM_NONBLOCK mode). - -If your program does nothing, but play sound, the easiest way is to put the device in PCM_NORMAL mode, and just -write as much data to the device as possible. This strategy can also be achieved by using a separate thread -with the sole task of playing out sound. - -In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few -periods by calling write a couple of times, and then use some timer method to write one period size of data to -the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer -doesn't expire exactly on time. - -Also note, that most timer API's that you can find for Python will cummulate time delays: If you set the timer -to expire after 1/10'th of a second, the actual timeout will happen slightly later, which will accumulate to -quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add -extra writes as nessecary. - -\subsection{Mixer Objects} -\label{mixer-objects} - -Mixer objects provides access to the ALSA mixer API. - -\begin{classdesc}{Mixer}{\optional{control}, \optional{id}, \optional{cardname}} -\var{control} - specifies which control to manipulate using this mixer object. The list -of available controls can be found with the \module{alsaaudio}.\function{mixers} function. -The default value is 'Master' - other common controls include 'Master Mono', 'PCM', 'Line', etc. - -\var{id} - the id of the mixer control. Default is 0 - -\var{cardname} - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -\end{classdesc} - -Mixer objects have the following methods: - -\begin{methoddesc}[Mixer]{cardname}{} -Return the name of the sound card used by this Mixer object -\end{methoddesc} - -\begin{methoddesc}[Mixer]{mixer}{} -Return the name of the specific mixer controlled by this object, For example 'Master' -or 'PCM' -\end{methoddesc} - -\begin{methoddesc}[Mixer]{mixerid}{} -Return the ID of the ALSA mixer controlled by this object. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{switchcap}{} -Returns a list of the switches which are defined by this specific mixer. Possible values in -this list are: - -\begin{tableii}{l|l}{Switches}{Switch}{Description} - \lineii{'Mute'}{This mixer can be muted} - \lineii{'Joined Mute'}{This mixer can mute all channels at the same time} - \lineii{'Playback Mute'}{This mixer can mute the playback output} - \lineii{'Joined Playback Mute'}{Mute playback for all channels at the same time} - \lineii{'Capture Mute'}{Mute sound capture} - \lineii{'Joined Capture Mute'}{Mute sound capture for all channels at a time} - \lineii{'Capture Exclusive'}{Not quite sure what this is} -\end{tableii} - -To manipulate these swithes use the \method{setrec} or \method{setmute} methods -\end{methoddesc} - -\begin{methoddesc}[Mixer]{volumecap}{} -Returns a list of the volume control capabilities of this mixer. Possible values in -the list are: - -\begin{tableii}{l|l}{Volume Capabilities}{Capability}{Description} - \lineii{'Volume'}{This mixer can control volume} - \lineii{'Joined Volume'}{This mixer can control volume for all channels at the same time} - \lineii{'Playback Volume'}{This mixer can manipulate the playback volume} - \lineii{'Joined Playback Volume'}{Manipulate playback volumne for all channels at the same time} - \lineii{'Capture Volume'}{Manipulate sound capture volume} - \lineii{'Joined Capture Volume'}{Manipulate sound capture volume for all channels at a time} -\end{tableii} - -\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. - -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]{getmute}{} -Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted. - -This method will fail if the mixer has no playback switch capabilities. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{getrec}{} -Return a list indicating the current record mute setting for each channel. 0 means not recording, 1 -means not recording. - -This method will fail if the mixer has no capture switch capabilities. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{setvolume}{volume,\optional{channel},\optional{direction}} -Change the current volume settings for this mixer. The \var{volume} argument controls -the new volume setting as an integer percentage. - -If the optional argument \var{channel} is present, the volume is set only for this channel. This -assumes that the mixer can control the volume for the channels independently. - -The optional \var{direction} argument can be either 'playback' or 'capture' is relevant if the mixer -has independent playback and capture volume capabilities, and controls which of the volumes -if changed. The default is 'playback' if the mixer has this capability, otherwise 'capture'. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{setmute}{mute, \optional{channel}} -Sets the mute flag to a new value. The \var{mute} argument is either 0 for not muted, or 1 for muted. - -The optional \var{channel} argument controls which channel is muted. The default is to set the mute flag -for all channels. - -This method will fail if the mixer has no playback mute capabilities -\end{methoddesc} - -\begin{methoddesc}[Mixer]{setrec}{capture,\optional{channel}} -Sets the capture mute flag to a new value. The \var{capture} argument is either 0 for no capture, -or 1 for capture. - -The optional \var{channel} argument controls which channel is changed. The default is to set the capture flag -for all channels. - -This method will fail if the mixer has no capture switch capabilities -\end{methoddesc} - - -\textbf{A Note on the ALSA Mixer API} - -The ALSA mixer API is extremely complicated - and hardly documented at all. \module{alsaaudio} implements -a much simplified way to access this API. In designing the API I've had to make some choices which -may limit what can and cannot be controlled through the API. However, If I had chosen to implement the -full API, I would have reexposed the horrible complexity/documentation ratio of the underlying API. -At least the \module{alsaaudio} API is easy to understand and use. - -If my design choises prevents you from doing something that the underlying API would have allowed, -please let me know, so I can incorporate these need into future versions. - -If the current state of affairs annoy you, the best you can do is to write a HOWTO on the API and -make this available on the net. Until somebody does this, the availability of ALSA mixer capable -devices will stay quite limited. - -Unfortunately, I'm not able to create such a HOWTO myself, since I only understand half of the API, -and that which I do understand has come from a painful trial and error process. - - - -% ==== 4. ==== -\subsection{ALSA Examples \label{pcm-example}} - -For now, the only examples available are the 'playbacktest.py' and the 'recordtest.py' programs included. -This will change in a future version. diff --git a/arch_src/pyalsaaudio-0.2/doc/src/.svn/text-base/pyalsaaudio.tex.svn-base b/arch_src/pyalsaaudio-0.2/doc/src/.svn/text-base/pyalsaaudio.tex.svn-base deleted file mode 100644 index 1a5a4b0..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/.svn/text-base/pyalsaaudio.tex.svn-base +++ /dev/null @@ -1,107 +0,0 @@ -\documentclass{howto} - -\title{PyAlsaAudio} - -\release{0.2} - -% At minimum, give your name and an email address. You can include a -% snail-mail address if you like. -\author{Casper Wilstrup} -\authoraddress{cwi@unispeed.com} - -\begin{document} -\maketitle - -% This makes the Abstract go on a separate page in the HTML version; -% if a copyright notice is used, it should go immediately after this. -% -\ifhtml -\chapter*{Front Matter\label{front}} -\fi - -% Copyright statement should go here, if needed. -This software is licensed under the PSF license - the same one used -by the majority of the python distribution. Basically you can use it -for anything you wish (even commercial purposes). There is no warranty -whatsoever. - -% The abstract should be a paragraph or two long, and describe the -% scope of the document. -\begin{abstract} -\noindent -This package contains wrappers for accessing the ALSA api from Python. It -is currently fairly complete for PCM devices and Mixer access. MIDI sequencer -support is low on my priority list, but volunteers are welcome. - -If you find bugs in the wrappers please notify me on email. Please -don't send bug reports regarding ALSA specifically. There are several -bugs in this api, and those should be reported to the ALSA team - not -me. -\end{abstract} - -\tableofcontents - -\section{What is ALSA} - -The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI -functionality to the Linux operating system. - -Logically ALSA consists of these components: -\begin{itemize} - \item A set of kernel drivers.\\ - These drivers are responsible for handling the physical sound - hardware from within the Linux kernel, and have been the standard - sound implementation in Linux since kernel version 2.5 - \item A kernel level API for manipulating the ALSA devices. - \item A user-space C library for simplified access to the sound hardware - from userspace applications. This library is called \textit{libasound} - and is required by all ALSA capable applications. -\end{itemize} - -More information about ALSA may be found on the project homepage -\url{http://www.alsa-project.org} - -\section{ALSA and Python} - -The older Linux sound API (OSS) which is now deprecated is well supported -from the standard Python library, through the ossaudiodev module. No native -ALSA support exists in the standard library (yet). - -There are a few other ``ALSA for Python'' projects available, including at -least two different projects called pyAlsa. Neither of these seem to be under -active development at the time - and neither are very feature complete. - -I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module -included in the standard Python library, but that is probably a while of yet. - -PyAlsaAudio hass full support for sound capture, playback of sound, as well as -the ALSA Mixer API. - -MIDI support is not available, and since I don't own any MIDI hardware, it's -difficult for me to implement it. Volunteers to work on this would be greatly -appreciated -\section{Installation} - -Note: the wrappers link with the alsasound library (from the alsa-lib -package). Verify that this is installed by looking for /usr/lib/libasound.so -before building. Naturally you also need to use a kernel with proper ALSA -support. This is the default in Linux kernel 2.6 and later. If you are using -kernel version 2.4 you may need to install the ALSA patches yourself - although -most distributions ship with ALSA kernels. - -To install, execute the following: \\ -\begin{verbatim} -$ python setup.py build -\end{verbatim} - -And then as root: \\ -\begin{verbatim} -# python setup.py install -\end{verbatim} - -\input{libalsaaudio} - - - - -\end{document} diff --git a/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/Makefile.svn-work b/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/Makefile.svn-work deleted file mode 100644 index 5af62dd..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/Makefile.svn-work +++ /dev/null @@ -1,5 +0,0 @@ -K 25 -svn:wc:ra_dav:version-url -V 66 -/svn/Shtoom/!svn/ver/1585/trunk/audio/pyalsaaudio/doc/src/Makefile -END diff --git a/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/libalsaaudio.tex.svn-work b/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/libalsaaudio.tex.svn-work deleted file mode 100644 index a30dac2..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/libalsaaudio.tex.svn-work +++ /dev/null @@ -1,5 +0,0 @@ -K 25 -svn:wc:ra_dav:version-url -V 74 -/svn/Shtoom/!svn/ver/1585/trunk/audio/pyalsaaudio/doc/src/libalsaaudio.tex -END diff --git a/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/pyalsaaudio.tex.svn-work b/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/pyalsaaudio.tex.svn-work deleted file mode 100644 index b1bf388..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/.svn/wcprops/pyalsaaudio.tex.svn-work +++ /dev/null @@ -1,5 +0,0 @@ -K 25 -svn:wc:ra_dav:version-url -V 73 -/svn/Shtoom/!svn/ver/1585/trunk/audio/pyalsaaudio/doc/src/pyalsaaudio.tex -END diff --git a/arch_src/pyalsaaudio-0.2/doc/src/Makefile b/arch_src/pyalsaaudio-0.2/doc/src/Makefile deleted file mode 100644 index 44c3313..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/Makefile +++ /dev/null @@ -1,20 +0,0 @@ -# Makefile for PyAlsaAudio documentation -# -# In order to use this you need an unpacked version of the -# Python source available. Set the source path below -# -# You also need a working latex installation, and the latex2html -# tool installed. -PYTHONSOURCE = /usr/src/Python-2.4.1/ - -# Shouldn't need to change anything below here! - -MKHOWTO = $(PYTHONSOURCE)/Doc/tools/mkhowto - -all: - $(MKHOWTO) --dir .. --html pyalsaaudio.tex - -text: - $(MKHOWTO) --dir .. --text pyalsaaudio.tex - - diff --git a/arch_src/pyalsaaudio-0.2/doc/src/libalsaaudio.tex b/arch_src/pyalsaaudio-0.2/doc/src/libalsaaudio.tex deleted file mode 100644 index c50ffe5..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/libalsaaudio.tex +++ /dev/null @@ -1,397 +0,0 @@ -\section{\module{alsaaudio}} - -%\declaremodule{builtin}{alsaaudio} % standard library, in C -\declaremodule{extension}{alsaaudio} % not standard, in C - -\platform{Linux} - -\moduleauthor{Casper Wilstrup}{cwi@unispeed.com} % Author of the module code; - - -\modulesynopsis{ALSA sound support} - - -The \module{alsaaudio} module defines functions and classes for using -ALSA. - -% ---- 3.1. ---- -% For each function, use a ``funcdesc'' block. This has exactly two -% parameters (each parameters is contained in a set of curly braces): -% the first parameter is the function name (this automatically -% generates an index entry); the second parameter is the function's -% argument list. If there are no arguments, use an empty pair of -% curly braces. If there is more than one argument, separate the -% arguments with backslash-comma. Optional parts of the parameter -% list are contained in \optional{...} (this generates a set of square -% brackets around its parameter). Arguments are automatically set in -% italics in the parameter list. Each argument should be mentioned at -% least once in the description; each usage (even inside \code{...}) -% should be enclosed in \var{...}. - -\begin{funcdesc}{mixers}{\optional{cardname}} -List the available mixers. The optional \var{cardname} specifies which -card should be queried (this is only relevant if you have more than one -sound card). Omit to use the default sound card -\end{funcdesc} - -\begin{classdesc}{PCM}{\optional{type}, \optional{mode}, \optional{cardname}} -This class is used to represent a PCM device (both playback and capture devices). -The arguments are: \\ -\var{type} - can be either PCM_CAPTURE or PCM_PLAYBACK (default). \\ -\var{mode} - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default).\\ -\var{cardname} - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -\end{classdesc} - -\begin{classdesc}{Mixer}{\optional{control}, \optional{id}, \optional{cardname}} -This class is used to access a specific ALSA mixer. -The arguments are: \\ -\var{control} - Name of the chosen mixed (default is Master). \\ -\var{id} - id of mixer (default is 0) -- More explaniation needed here\\ -\var{cardname} specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -\end{classdesc} - - -\begin{excdesc}{ALSAAudioError} -Exception raised when an operation fails for a ALSA specific reason. -The exception argument is a string describing the reason of the -failure. -\end{excdesc} - -\subsection{PCM Terminology and Concepts} - -In order to use PCM devices it is useful to be familiar with some concepts and -terminology. - -\begin{description} -\item[Sample] PCM audio, whether it is input or output, consists at the lowest level -of a number of single samples. A sample represents the sound in a single channel in -a brief interval. If more than one channel is in use, more than one sample is required -for each interval to describe the sound. Samples can be of many different sizes, ranging -from 8 bit to 64 bit presition. The specific format of each sample can also vary - they -can be big endian byte order, little endian byte order, or even floats. - -\item[Frame] A frame consists of exactly one sample per channel. If there is only one -channel (Mono sound) a frame is simply a single sample. If the sound is stereo, each frame -consists of two samples, etc. - -\item[Frame size] This is the size in bytes of each frame. This can vary a lot: if each sample is -8 bits, and we're handling mono sound, the frame size is one byte. Similarly in 6 channel audio with -64 bit floating point samples, the frame size is 48 bytes - -\item[Rate] PCM sound consists of a flow of sound frames. The sound rate controls how often -the current frame is replaced. For example, a rate of 8000 Hz means that a new frame is played -or captured 8000 times per second. - -\item[Data rate] This is the number of bytes, which must be recorded or provided per second -at a certain frame size and rate. - -8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s - -At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit (8 bytes) samples -has a data rate of 96000 * 6 * 8 = 4608 kb/s (almost 5 Mb sound data per second) - -\item[Period] When the hardware processes data this is done in chunks of frames. The time interval -between each processing (A/D or D/A conversion) is known as the period. The size of the period has -direct implication on the latency of the sound input or output. For low-latency the period size should -be very small, while low CPU resource usage would usually demand larger period sizes. With ALSA, the -CPU utilization is not impacted much by the period size, since the kernel layer buffers multiple -periods internally, so each period generates an interrupt and a memory copy, but userspace can be -slower and read or write multiple periods at the same time. - -\item[Period size] This is the size of each period in Hz. \emph{Not bytes, but Hz!.} In \module{alsaaudio} -the period size is set directly, and it is therefore important to understand the significance of this -number. If the period size is configured to for example 32, each write should contain exactly 32 frames -of sound data, and each read will return either 32 frames of data or nothing at all. - -\end{description} - -Once you understand these concepts, you will be ready to actually utilize PCM API. Read on. - -\subsection{PCM Objects} -\label{pcm-objects} - -The acronym PCM is short for Pulse Code Modulation and is the method used in ALSA -and many other places to handle playback and capture of sampled sound data. - -PCM objects in \module{alsaaudio} are used to do exactly that, either play sample based -sound or capture sound from some input source (perhaps a microphone). The PCM object -constructor takes the following arguments: - -\begin{classdesc}{PCM}{\optional{type}, \optional{mode}, \optional{cardname}} - -\var{type} - can be either PCM_CAPTURE or PCM_PLAYBACK (default). - -\var{mode} - can be either PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL (the default). -In PCM_NONBLOCK mode, calls to read will return immediately independent of wether -there is any actual data to read. Similarly, write calls will return immediately -without actually writing anything to the playout buffer if the buffer is full. - -In the current version of \module{alsaaudio} PCM_ASYNC is useless, since it relies -on a callback procedure, which can't be specified from Python. - -\var{cardname} - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card - -This will construct a PCM object with default settings: - -Sample format: PCM_FORMAT_S16_LE \\ -Rate: 8000 Hz \\ -Channels: 2 \\ -Period size: 32 frames \\ -\end{classdesc} - -PCM objects have the following methods: - -\begin{methoddesc}[PCM]{pcmtype}{} -Returns the type of PCM object. Either PCM_CAPTURE or PCM_PLAYBACK. -\end{methoddesc} - -\begin{methoddesc}[PCM]{pcmmode}{} -Return the mode of the PCM object. One of PCM_NONBLOCK, PCM_ASYNC, or PCM_NORMAL -\end{methoddesc} - -\begin{methoddesc}[PCM]{cardname}{} -Return the name of the sound card used by this PCM object. -\end{methoddesc} - -\begin{methoddesc}[PCM]{setchannels}{nchannels} -Used to set the number of capture or playback channels. Common values are: 1 = mono, 2 = stereo, -and 6 = full 6 channel audio. Few sound cards support more than 2 channels -\end{methoddesc} - -\begin{methoddesc}[PCM]{setrate}{rate} -Set the sample rate in Hz for the device. Typical values are 8000 (poor sound), 16000, 44100 (cd quality), -and 96000 -\end{methoddesc} - -\begin{methoddesc}[PCM]{setformat}{} -The sound format of the device. Sound format controls how the PCM device interpret data for playback, -and how data is encoded in captures. - -The following formats are provided by ALSA: -\begin{tableii}{l|l}{Formats}{Format}{Description} - \lineii{PCM_FORMAT_S8}{Signed 8 bit samples for each channel} - \lineii{PCM_FORMAT_U8}{Signed 8 bit samples for each channel} - \lineii{PCM_FORMAT_S16_LE}{Signed 16 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_S16_BE}{Signed 16 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_U16_LE}{Unsigned 16 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_U16_BE}{Unsigned 16 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_S24_LE}{Signed 24 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_S24_BE}{Signed 24 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_U24_LE}{Unsigned 24 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_U24_BE}{Unsigned 24 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_S32_LE}{Signed 32 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_S32_BE}{Signed 32 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_U32_LE}{Unsigned 32 bit samples for each channel (Little Endian byte order)} - \lineii{PCM_FORMAT_U32_BE}{Unsigned 32 bit samples for each channel (Big Endian byte order)} - \lineii{PCM_FORMAT_FLOAT_LE}{32 bit samples encoded as float. (Little Endian byte order)} - \lineii{PCM_FORMAT_FLOAT_BE}{32 bit samples encoded as float (Big Endian byte order)} - \lineii{PCM_FORMAT_FLOAT64_LE}{64 bit samples encoded as float. (Little Endian byte order)} - \lineii{PCM_FORMAT_FLOAT64_BE}{64 bit samples encoded as float. (Big Endian byte order)} - \lineii{PCM_FORMAT_MU_LAW}{A logarithmic encoding (used by Sun .au files)} - \lineii{PCM_FORMAT_A_LAW}{Another logarithmic encoding} - \lineii{PCM_FORMAT_IMA_ADPCM}{a 4:1 compressed format defined by the Interactive Multimedia Association} - \lineii{PCM_FORMAT_MPEG}{MPEG encoded audio?} - \lineii{PCM_FORMAT_GSM}{9600 constant rate encoding well suitet for speech} -\end{tableii} - -\end{methoddesc} - -\begin{methoddesc}[PCM]{setperiodsize}{period} -Sets the actual period size in frames. Each write should consist of exactly this number of frames, and -each read will return this number of frames (unless the device is in PCM_NONBLOCK mode, in which case -it may return nothing at all) -\end{methoddesc} - -\begin{methoddesc}[PCM]{read}{} -In PCM_NORMAL mode, this function blocks until a full period is available, and then returns a -tuple (length,data) where \emph{length} is the size in bytes of the captured data, and \emph{data} -is the captured sound frames as a string. The length of the returned data will be periodsize*framesize -bytes. - -In PCM_NONBLOCK mode, the call will not block, but will return \code{(0,'')} if no new period -has become available since the last call to read. -\end{methoddesc} - -\begin{methoddesc}[PCM]{write}{data} -Writes (plays) the sound in data. The length of data \emph{must} be a multiple of the frame size, and -\emph{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 PCM_NONBLOCK mode, this call will block if the kernel buffer is full, and -until enough sound has been played to allow the sound data to be buffered. The call always returns -the size of the data provided - -In PCM_NONBLOCK mode, the call will return immediately, with a return value of zero, if the buffer is -full. In this case, the data should be written at a later time. - -\end{methoddesc} - -\strong{A few hints on using PCM devices for playback} - -The most common reason for problems with playback of PCM audio, is that the people don't properly understand -that writes to PCM devices must match \emph{exactly} 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 written to the device, the write function will either block (PCM_NORMAL mode) or return zero -(PCM_NONBLOCK mode). - -If your program does nothing, but play sound, the easiest way is to put the device in PCM_NORMAL mode, and just -write as much data to the device as possible. This strategy can also be achieved by using a separate thread -with the sole task of playing out sound. - -In GUI programs, however, it may be a better strategy to setup the device, preload the buffer with a few -periods by calling write a couple of times, and then use some timer method to write one period size of data to -the device every period. The purpose of the preloading is to avoid underrun clicks if the used timer -doesn't expire exactly on time. - -Also note, that most timer API's that you can find for Python will cummulate time delays: If you set the timer -to expire after 1/10'th of a second, the actual timeout will happen slightly later, which will accumulate to -quite a lot after a few seconds. Hint: use time.time() to check how much time has really passed, and add -extra writes as nessecary. - -\subsection{Mixer Objects} -\label{mixer-objects} - -Mixer objects provides access to the ALSA mixer API. - -\begin{classdesc}{Mixer}{\optional{control}, \optional{id}, \optional{cardname}} -\var{control} - specifies which control to manipulate using this mixer object. The list -of available controls can be found with the \module{alsaaudio}.\function{mixers} function. -The default value is 'Master' - other common controls include 'Master Mono', 'PCM', 'Line', etc. - -\var{id} - the id of the mixer control. Default is 0 - -\var{cardname} - specifies which card should be used (this is only relevant -if you have more than one sound card). Omit to use the default sound card -\end{classdesc} - -Mixer objects have the following methods: - -\begin{methoddesc}[Mixer]{cardname}{} -Return the name of the sound card used by this Mixer object -\end{methoddesc} - -\begin{methoddesc}[Mixer]{mixer}{} -Return the name of the specific mixer controlled by this object, For example 'Master' -or 'PCM' -\end{methoddesc} - -\begin{methoddesc}[Mixer]{mixerid}{} -Return the ID of the ALSA mixer controlled by this object. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{switchcap}{} -Returns a list of the switches which are defined by this specific mixer. Possible values in -this list are: - -\begin{tableii}{l|l}{Switches}{Switch}{Description} - \lineii{'Mute'}{This mixer can be muted} - \lineii{'Joined Mute'}{This mixer can mute all channels at the same time} - \lineii{'Playback Mute'}{This mixer can mute the playback output} - \lineii{'Joined Playback Mute'}{Mute playback for all channels at the same time} - \lineii{'Capture Mute'}{Mute sound capture} - \lineii{'Joined Capture Mute'}{Mute sound capture for all channels at a time} - \lineii{'Capture Exclusive'}{Not quite sure what this is} -\end{tableii} - -To manipulate these swithes use the \method{setrec} or \method{setmute} methods -\end{methoddesc} - -\begin{methoddesc}[Mixer]{volumecap}{} -Returns a list of the volume control capabilities of this mixer. Possible values in -the list are: - -\begin{tableii}{l|l}{Volume Capabilities}{Capability}{Description} - \lineii{'Volume'}{This mixer can control volume} - \lineii{'Joined Volume'}{This mixer can control volume for all channels at the same time} - \lineii{'Playback Volume'}{This mixer can manipulate the playback volume} - \lineii{'Joined Playback Volume'}{Manipulate playback volumne for all channels at the same time} - \lineii{'Capture Volume'}{Manipulate sound capture volume} - \lineii{'Joined Capture Volume'}{Manipulate sound capture volume for all channels at a time} -\end{tableii} - -\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. - -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]{getmute}{} -Return a list indicating the current mute setting for each channel. 0 means not muted, 1 means muted. - -This method will fail if the mixer has no playback switch capabilities. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{getrec}{} -Return a list indicating the current record mute setting for each channel. 0 means not recording, 1 -means not recording. - -This method will fail if the mixer has no capture switch capabilities. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{setvolume}{volume,\optional{channel},\optional{direction}} -Change the current volume settings for this mixer. The \var{volume} argument controls -the new volume setting as an integer percentage. - -If the optional argument \var{channel} is present, the volume is set only for this channel. This -assumes that the mixer can control the volume for the channels independently. - -The optional \var{direction} argument can be either 'playback' or 'capture' is relevant if the mixer -has independent playback and capture volume capabilities, and controls which of the volumes -if changed. The default is 'playback' if the mixer has this capability, otherwise 'capture'. -\end{methoddesc} - -\begin{methoddesc}[Mixer]{setmute}{mute, \optional{channel}} -Sets the mute flag to a new value. The \var{mute} argument is either 0 for not muted, or 1 for muted. - -The optional \var{channel} argument controls which channel is muted. The default is to set the mute flag -for all channels. - -This method will fail if the mixer has no playback mute capabilities -\end{methoddesc} - -\begin{methoddesc}[Mixer]{setrec}{capture,\optional{channel}} -Sets the capture mute flag to a new value. The \var{capture} argument is either 0 for no capture, -or 1 for capture. - -The optional \var{channel} argument controls which channel is changed. The default is to set the capture flag -for all channels. - -This method will fail if the mixer has no capture switch capabilities -\end{methoddesc} - - -\textbf{A Note on the ALSA Mixer API} - -The ALSA mixer API is extremely complicated - and hardly documented at all. \module{alsaaudio} implements -a much simplified way to access this API. In designing the API I've had to make some choices which -may limit what can and cannot be controlled through the API. However, If I had chosen to implement the -full API, I would have reexposed the horrible complexity/documentation ratio of the underlying API. -At least the \module{alsaaudio} API is easy to understand and use. - -If my design choises prevents you from doing something that the underlying API would have allowed, -please let me know, so I can incorporate these need into future versions. - -If the current state of affairs annoy you, the best you can do is to write a HOWTO on the API and -make this available on the net. Until somebody does this, the availability of ALSA mixer capable -devices will stay quite limited. - -Unfortunately, I'm not able to create such a HOWTO myself, since I only understand half of the API, -and that which I do understand has come from a painful trial and error process. - - - -% ==== 4. ==== -\subsection{ALSA Examples \label{pcm-example}} - -For now, the only examples available are the 'playbacktest.py' and the 'recordtest.py' programs included. -This will change in a future version. diff --git a/arch_src/pyalsaaudio-0.2/doc/src/pyalsaaudio.tex b/arch_src/pyalsaaudio-0.2/doc/src/pyalsaaudio.tex deleted file mode 100644 index 1a5a4b0..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/src/pyalsaaudio.tex +++ /dev/null @@ -1,107 +0,0 @@ -\documentclass{howto} - -\title{PyAlsaAudio} - -\release{0.2} - -% At minimum, give your name and an email address. You can include a -% snail-mail address if you like. -\author{Casper Wilstrup} -\authoraddress{cwi@unispeed.com} - -\begin{document} -\maketitle - -% This makes the Abstract go on a separate page in the HTML version; -% if a copyright notice is used, it should go immediately after this. -% -\ifhtml -\chapter*{Front Matter\label{front}} -\fi - -% Copyright statement should go here, if needed. -This software is licensed under the PSF license - the same one used -by the majority of the python distribution. Basically you can use it -for anything you wish (even commercial purposes). There is no warranty -whatsoever. - -% The abstract should be a paragraph or two long, and describe the -% scope of the document. -\begin{abstract} -\noindent -This package contains wrappers for accessing the ALSA api from Python. It -is currently fairly complete for PCM devices and Mixer access. MIDI sequencer -support is low on my priority list, but volunteers are welcome. - -If you find bugs in the wrappers please notify me on email. Please -don't send bug reports regarding ALSA specifically. There are several -bugs in this api, and those should be reported to the ALSA team - not -me. -\end{abstract} - -\tableofcontents - -\section{What is ALSA} - -The Advanced Linux Sound Architecture (ALSA) provides audio and MIDI -functionality to the Linux operating system. - -Logically ALSA consists of these components: -\begin{itemize} - \item A set of kernel drivers.\\ - These drivers are responsible for handling the physical sound - hardware from within the Linux kernel, and have been the standard - sound implementation in Linux since kernel version 2.5 - \item A kernel level API for manipulating the ALSA devices. - \item A user-space C library for simplified access to the sound hardware - from userspace applications. This library is called \textit{libasound} - and is required by all ALSA capable applications. -\end{itemize} - -More information about ALSA may be found on the project homepage -\url{http://www.alsa-project.org} - -\section{ALSA and Python} - -The older Linux sound API (OSS) which is now deprecated is well supported -from the standard Python library, through the ossaudiodev module. No native -ALSA support exists in the standard library (yet). - -There are a few other ``ALSA for Python'' projects available, including at -least two different projects called pyAlsa. Neither of these seem to be under -active development at the time - and neither are very feature complete. - -I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module -included in the standard Python library, but that is probably a while of yet. - -PyAlsaAudio hass full support for sound capture, playback of sound, as well as -the ALSA Mixer API. - -MIDI support is not available, and since I don't own any MIDI hardware, it's -difficult for me to implement it. Volunteers to work on this would be greatly -appreciated -\section{Installation} - -Note: the wrappers link with the alsasound library (from the alsa-lib -package). Verify that this is installed by looking for /usr/lib/libasound.so -before building. Naturally you also need to use a kernel with proper ALSA -support. This is the default in Linux kernel 2.6 and later. If you are using -kernel version 2.4 you may need to install the ALSA patches yourself - although -most distributions ship with ALSA kernels. - -To install, execute the following: \\ -\begin{verbatim} -$ python setup.py build -\end{verbatim} - -And then as root: \\ -\begin{verbatim} -# python setup.py install -\end{verbatim} - -\input{libalsaaudio} - - - - -\end{document} diff --git a/arch_src/pyalsaaudio-0.2/doc/up.gif b/arch_src/pyalsaaudio-0.2/doc/up.gif deleted file mode 100644 index a9d3e13..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/up.gif +++ /dev/null Binary files differ diff --git a/arch_src/pyalsaaudio-0.2/doc/up.png b/arch_src/pyalsaaudio-0.2/doc/up.png deleted file mode 100644 index a90e028..0000000 --- a/arch_src/pyalsaaudio-0.2/doc/up.png +++ /dev/null Binary files differ diff --git a/arch_src/pyalsaaudio-0.2/mixertest.py b/arch_src/pyalsaaudio-0.2/mixertest.py deleted file mode 100644 index 642ae10..0000000 --- a/arch_src/pyalsaaudio-0.2/mixertest.py +++ /dev/null @@ -1,88 +0,0 @@ -## mixertest.py -## -## This is an example of using the ALSA mixer API -## -## The script will set the volume or mute switch of the specified Mixer -## depending on command line options. -## -## Examples: -## python mixertest.py # list available mixers -## python mixertest.py Master # show Master mixer settings -## python mixertest.py Master 80 # set the master volume to 80% -## python mixertest.py Master 1,90 # set channel 1 volume to 90% -## python mixertest.py Master mute # mute the master mixer -## python mixertest.py Master unmute # unmute the master mixer - -import alsaaudio -import sys - -if len(sys.argv) == 1: - # Demonstrates how to read the available mixers - print "Available mixer controls:" - for m in alsaaudio.mixers(): - print " '%s'" % m - -if len(sys.argv) == 2: - # Demonstrates how mixer settings are queried. - name = sys.argv[1] - try: - mixer = alsaaudio.Mixer(name) - except alsaaudio.ALSAAudioError: - print "No such mixer" - sys.exit(1) - - print "Mixer name: '%s'"%mixer.mixer() - print "Capabilities",mixer.volumecap()+mixer.switchcap() - volumes = mixer.getvolume() - for i in range(len(volumes)): - print "Channel %i volume: %i%%"%(i,volumes[i]) - - try: - mutes = mixer.getmute() - for i in range(len(mutes)): - if mutes[i]: print "Channel %i is muted"%i - except alsaaudio.ALSAAudioError: - # May not support muting - pass - - try: - recs = mixer.getrec() - for i in range(len(recs)): - if recs[i]: print "Channel %i is recording"%i - except alsaaudio.ALSAAudioError: - # May not support recording - pass - -if (len(sys.argv)) == 3: - # Demonstrates how to set mixer settings - name = sys.argv[1] - try: - mixer = alsaaudio.Mixer(name) - except alsaaudio.ALSAAudioError: - print "No such mixer" - sys.exit(1) - - args = sys.argv[2] - if args in ['mute','unmute']: - # Mute/unmute the mixer - if args == 'mute': mixer.setmute(1) - else: mixer.setmute(0) - sys.exit(0) - if args in ['rec','unrec']: - # Enable/disable recording - if args == 'rec': mixer.setrec(1) - else: mixer.setrec(0) - sys.exit(0) - - - if args.find(',')!=-1: - channel,volume = map(int,args.split(',')) - else: - channel = alsaaudio.MIXER_CHANNEL_ALL - volume = int(args) - # Set volume for specified channel. MIXER_CHANNEL_ALL means set - # volume for all channels - mixer.setvolume(volume,channel) - - - diff --git a/arch_src/pyalsaaudio-0.2/modlibalsaaudio.ind b/arch_src/pyalsaaudio-0.2/modlibalsaaudio.ind deleted file mode 100644 index 8af6c8a..0000000 --- a/arch_src/pyalsaaudio-0.2/modlibalsaaudio.ind +++ /dev/null @@ -1,4 +0,0 @@ -\ -\begin{theindex} -\label{modindex} -\end{theindex} diff --git a/arch_src/pyalsaaudio-0.2/playbacktest.py b/arch_src/pyalsaaudio-0.2/playbacktest.py deleted file mode 100644 index 06e0fe9..0000000 --- a/arch_src/pyalsaaudio-0.2/playbacktest.py +++ /dev/null @@ -1,36 +0,0 @@ -## recordtest.py -## -## This is an example of a simple sound playback script. -## -## The script opens an ALSA pcm for sound playback. Set -## various attributes of the device. It then reads data -## from stdin and writes it to the device. -## -## To test it out do the following: -## python recordtest.py > out.raw # talk to the microphone -## python playbacktest.py < out.raw -## -## If you have Gnome, you could also just test by doing something like: -## python playbacktest.py < /usr/share/sounds/gnibbles/laughter.wav -import alsaaudio -import sys -import time - -# Open the device in playback mode. -out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK) - -# Set attributes: Mono, 8000 Hz, 16 bit little endian frames -out.setchannels(1) -out.setrate(8000) -out.setformat(alsaaudio.PCM_FORMAT_S16_LE) - -# The period size controls the internal number of frames per period. -# The significance of this parameter is documented in the ALSA api. -out.setperiodsize(160) - -loops = 10000 -while loops > 0: - loops -= 1 - # Read data from stdin - data = sys.stdin.read(320) - out.write(data) diff --git a/arch_src/pyalsaaudio-0.2/recordtest.py b/arch_src/pyalsaaudio-0.2/recordtest.py deleted file mode 100644 index 216d627..0000000 --- a/arch_src/pyalsaaudio-0.2/recordtest.py +++ /dev/null @@ -1,45 +0,0 @@ -## recordtest.py -## -## This is an example of a simple sound capture script. -## -## The script opens an ALSA pcm forsound capture. Set -## various attributes of the capture, and reads in a loop, -## writing the data to standard out. -## -## To test it out do the following: -## python recordtest.py > out.raw # talk to the microphone -## aplay -r 8000 -f S16_LE -c 1 out.raw - -import alsaaudio -import sys -import time - -# Open the device in nonblocking capture mode. The last argument could -# just as well have been zero for blocking mode. Then we could have -# left out the sleep call in the bottom of the loop -inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE,alsaaudio.PCM_NONBLOCK) - -# Set attributes: Mono, 8000 Hz, 16 bit little endian samples -inp.setchannels(1) -inp.setrate(8000) -inp.setformat(alsaaudio.PCM_FORMAT_S16_LE) - -# The period size controls the internal number of frames per period. -# The significance of this parameter is documented in the ALSA api. -# For our purposes, it is suficcient to know that reads from the device -# will return this many frames. Each frame being 2 bytes long. -# 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.setperiodsize(160) - -loops = 1000000 -while loops > 0: - loops -= 1 - # Read data from device - l,data = inp.read() - - if l: - # actual data read. Write it to stdout - sys.stdout.write(data) - time.sleep(.001) diff --git a/arch_src/pyalsaaudio-0.2/setup.py b/arch_src/pyalsaaudio-0.2/setup.py deleted file mode 100644 index d2534a7..0000000 --- a/arch_src/pyalsaaudio-0.2/setup.py +++ /dev/null @@ -1,14 +0,0 @@ -from distutils.core import setup -from distutils.extension import Extension - -setup( - name = "alsaaudio", - version = "0.1", - description = "alsa bindings", - author = "Casper Wilstrup", - author_email="cwi@unispeed.com", - ext_modules=[Extension("alsaaudio",["alsaaudio.c"],libraries=['asound']) - ] - ) - - diff --git a/arch_src/pyalsaaudio-0.2/test.py b/arch_src/pyalsaaudio-0.2/test.py deleted file mode 100644 index 51707ec..0000000 --- a/arch_src/pyalsaaudio-0.2/test.py +++ /dev/null @@ -1,7 +0,0 @@ -import alsaaudio -import sys -if len(sys.argv) > 1: name = sys.argv[1] -else: name = "Master" - -m = alsaaudio.Mixer(name) - diff --git a/arch_src/pyalsaaudio_0.2-1ubuntu1.dsc b/arch_src/pyalsaaudio_0.2-1ubuntu1.dsc deleted file mode 100644 index a8f327b..0000000 --- a/arch_src/pyalsaaudio_0.2-1ubuntu1.dsc +++ /dev/null @@ -1,27 +0,0 @@ ------BEGIN PGP SIGNED MESSAGE----- -Hash: RIPEMD160 - -Format: 1.0 -Source: pyalsaaudio -Version: 0.2-1ubuntu1 -Binary: python-alsaaudio -Maintainer: Ubuntu MOTU Developers -Architecture: any -Standards-Version: 3.7.2 -Build-Depends: debhelper (>= 5.0.37.2), python-all-dev (>= 2.3.5-11), python-support (>= 0.3), libasound2-dev -Python-Version: all -Original-Maintainer: Florian Ragwitz -Uploaders: Paul Brossier -Files: - 23163f572f8f3a0b295f48f90165e767 88819 pyalsaaudio_0.2.orig.tar.gz - 8170b2430e2d836521110ceaa76089b4 3031 pyalsaaudio_0.2-1ubuntu1.diff.gz - ------BEGIN PGP SIGNATURE----- -Version: GnuPG v2.0.3 (GNU/Linux) - -iQCVAwUBRgMP+aPP1313boLqAQPoDAP7BbkyJT7xEXsxLvoPvrA3j+xC0Hv/qF1s -Q80rU9WTKC9jJCqNcG4evg3u7p7amlqyCt6OAB5gajZBxSMhw4FG5B/D/XcxrWp0 -q32anQEC9JVpan6wYLPvb4D1bid6Gx9A3cgyePbsv/3cfUWzghXYLNn8vom0/VSd -EwoJzNjNyRY= -=iwYv ------END PGP SIGNATURE----- diff --git a/bin/compile-archspec b/bin/compile-archspec deleted file mode 100755 index 2a5a042..0000000 --- a/bin/compile-archspec +++ /dev/null @@ -1,16 +0,0 @@ -#!/bin/bash -set -e - -archsrcdir="$1" -archinstdir="$2" - -mkdir -p "${archinstdir}" -cp -a "${archsrcdir}"/* "${archinstdir}"/ -cd "${archinstdir}" -if [ -e configure ] ; then - ./configure --prefix="${archinstdir}" -fi - -make PREFIX="${archinstdir}" -make install PREFIX="${archinstdir}" - diff --git a/bin/wrapper b/bin/wrapper deleted file mode 100755 index 0d92661..0000000 --- a/bin/wrapper +++ /dev/null @@ -1,22 +0,0 @@ -#!/bin/bash -set -e -instflag="${SUGAR_ACTIVITY_ROOT}/data/.installed" -cpu="$(uname -m)" -archsrcdir="${SUGAR_BUNDLE_PATH}/arch_src" -archbundledir="${SUGAR_BUNDLE_PATH}/arch_${cpu}" -archinstdir="${SUGAR_ACTIVITY_ROOT}/data/.arch_${cpu}" -pythonversion="$(python -V 2>&1 | cut -d ' ' -f 2 | cut -d . -f 1-2)" - -if [ ! -e "${instflag}" ] ; then - curchecksum="$(find "${archsrcdir}" -type f -print0 | xargs -0 md5sum -- | md5sum)" - if [ "$cpu" = "unknown" -o ! -e "${archbundledir}" ] || [ "${curchecksum}" != "$(cat "${archbundledir}/checksum")" ] ; then - # unknown arch or no (current) precompiled binaries -> need to compile - compile-archspec "${archsrcdir}" "${archinstdir}" - fi - touch "${instflag}" -fi - -export LD_LIBRARY_PATH="${archinstdir}/lib:${archinstdir}/lib64:${LD_LIBRARY_PATH}" -export PYTHONPATH="${archinstdir}/lib/python${pythonversion}/site-packages:${PYTHONPATH}" - -exec "$@" -- cgit v0.9.1