Merge branch 'master' into turbo

17 files changed, 0 insertions, 2090 deletions

diff --git a/gr-audio-alsa/.gitignore b/gr-audio-alsa/.gitignore
deleted file mode 100644
index cdcf41b15..000000000
--- a/gr-audio-alsa/.gitignore
+++ /dev/null
@@ -1,30 +0,0 @@
-/*.cache
-/*.la
-/*.lo
-/*.pc
-/.deps
-/.la
-/.libs
-/.lo
-/Makefile
-/Makefile.in
-/aclocal.m4
-/autom4te.cache
-/config.cache
-/config.h
-/config.h.in
-/config.log
-/config.status
-/configure
-/depcomp
-/install-sh
-/libtool
-/ltmain.sh
-/make.log
-/missing
-/missing
-/mkinstalldirs
-/py-compile
-/stamp-h
-/stamp-h.in
-/stamp-h1
diff --git a/gr-audio-alsa/Makefile.am b/gr-audio-alsa/Makefile.am
deleted file mode 100644
index 5bd92b0ff..000000000
--- a/gr-audio-alsa/Makefile.am
+++ /dev/null
@@ -1,30 +0,0 @@
-#
-# Copyright 2004,2006,2009 Free Software Foundation, Inc.
-# 
-# This file is part of GNU Radio
-# 
-# GNU Radio is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 3, or (at your option)
-# any later version.
-# 
-# GNU Radio is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-# 
-# You should have received a copy of the GNU General Public License
-# along with GNU Radio; see the file COPYING.  If not, write to
-# the Free Software Foundation, Inc., 51 Franklin Street,
-# Boston, MA 02110-1301, USA.
-# 
-
-include $(top_srcdir)/Makefile.common
-
-SUBDIRS = src
-
-pkgconfigdir = $(libdir)/pkgconfig
-dist_pkgconfig_DATA = gnuradio-audio-alsa.pc
-
-etcdir = $(gr_prefsdir)
-dist_etc_DATA = gr-audio-alsa.conf
diff --git a/gr-audio-alsa/build-stamp b/gr-audio-alsa/build-stamp
deleted file mode 100644
index e69de29bb..000000000
--- a/gr-audio-alsa/build-stamp
+++ /dev/null
diff --git a/gr-audio-alsa/gnuradio-audio-alsa.pc.in b/gr-audio-alsa/gnuradio-audio-alsa.pc.in
deleted file mode 100644
index d5147e6af..000000000
--- a/gr-audio-alsa/gnuradio-audio-alsa.pc.in
+++ /dev/null
@@ -1,11 +0,0 @@
-prefix=@prefix@
-exec_prefix=@exec_prefix@
-libdir=@libdir@
-includedir=@includedir@
-
-Name: gnuradio-audio-alsa
-Description: The GNU Radio block for the ALSA sound system
-Requires: gnuradio-core alsa
-Version: @LIBVER@
-Libs: -L${libdir} -lgnuradio-audio-alsa
-Cflags: -I${includedir}
diff --git a/gr-audio-alsa/gr-audio-alsa.conf b/gr-audio-alsa/gr-audio-alsa.conf
deleted file mode 100644
index 5cec63e7a..000000000
--- a/gr-audio-alsa/gr-audio-alsa.conf
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file contains system wide configuration data for GNU Radio.
-# You may override any setting on a per-user basis by editing
-# ~/.gnuradio/config.conf
-
-[audio_alsa]
-
-default_input_device = hw:0,0
-default_output_device = hw:0,0
-period_time = 0.010                      # in seconds
-nperiods = 4				 # total buffering = period_time * nperiods
-verbose = false
diff --git a/gr-audio-alsa/src/.gitignore b/gr-audio-alsa/src/.gitignore
deleted file mode 100644
index b751c6731..000000000
--- a/gr-audio-alsa/src/.gitignore
+++ /dev/null
@@ -1,16 +0,0 @@
-/Makefile
-/Makefile.in
-/.la
-/.lo
-/.deps
-/.libs
-/*.la
-/*.lo
-/usrp.py
-/usrp.cc
-/audio_oss.cc
-/audio_oss.py
-/audio_alsa.py
-/audio_alsa.cc
-/run_tests
-/*.pyc
diff --git a/gr-audio-alsa/src/Makefile.am b/gr-audio-alsa/src/Makefile.am
deleted file mode 100644
index ed92cc197..000000000
--- a/gr-audio-alsa/src/Makefile.am
+++ /dev/null
@@ -1,82 +0,0 @@
-#
-# Copyright 2004,2008,2009 Free Software Foundation, Inc.
-# 
-# This file is part of GNU Radio
-# 
-# GNU Radio is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 3, or (at your option)
-# any later version.
-# 
-# GNU Radio is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-# 
-# You should have received a copy of the GNU General Public License
-# along with GNU Radio; see the file COPYING.  If not, write to
-# the Free Software Foundation, Inc., 51 Franklin Street,
-# Boston, MA 02110-1301, USA.
-# 
-
-include $(top_srcdir)/Makefile.common
-
-EXTRA_DIST = run_tests.in
-
-# C/C++ headers get installed in ${prefix}/include/gnuradio
-grinclude_HEADERS =			\
-	audio_alsa_sink.h		\
-	audio_alsa_source.h
-
-noinst_HEADERS = 			\
-	gri_alsa.h
-
-noinst_PYTHON = 			\
-	qa_alsa.py
-
-AM_CPPFLAGS = $(STD_DEFINES_AND_INCLUDES) $(PYTHON_CPPFLAGS) $(WITH_INCLUDES)
-
-###################################
-# ALSA library, including OS interface, sink, and source
-
-lib_LTLIBRARIES = libgnuradio-audio-alsa.la
-
-libgnuradio_audio_alsa_la_SOURCES = 	\
-	audio_alsa_sink.cc		\
-	audio_alsa_source.cc		\
-	gri_alsa.cc
-
-libgnuradio_audio_alsa_la_LIBADD =	\
-	$(GNURADIO_CORE_LA)		\
-	$(ALSA_LIBS)
-
-libgnuradio_audio_alsa_la_LDFLAGS = $(NO_UNDEFINED) $(LTVERSIONFLAGS)
-
-if PYTHON
-###################################
-# SWIG Python interface and library
-TESTS = run_tests
-DISTCLEANFILES = run_tests
-
-TOP_SWIG_IFILES = 			\
-	audio_alsa.i
-
-# Install so that they end up available as:
-#   import gnuradio.audio_alsa
-# This ends up at:
-#   ${prefix}/lib/python${python_version}/site-packages/gnuradio
-audio_alsa_pythondir_category =		\
-	gnuradio
-
-# additional libraries for linking with the SWIG-generated library
-audio_alsa_la_swig_libadd =		\
-	libgnuradio-audio-alsa.la
-
-include $(top_srcdir)/Makefile.swig
-
-# add some of the variables generated inside the Makefile.swig.gen
-BUILT_SOURCES = $(swig_built_sources)
-
-# Do not distribute the output of SWIG
-no_dist_files = $(swig_built_sources)
-endif
\ No newline at end of file
diff --git a/gr-audio-alsa/src/Makefile.swig.gen b/gr-audio-alsa/src/Makefile.swig.gen
deleted file mode 100644
index 3a84a02b4..000000000
--- a/gr-audio-alsa/src/Makefile.swig.gen
+++ /dev/null
@@ -1,259 +0,0 @@
-# -*- Makefile -*-
-#
-# Copyright 2009 Free Software Foundation, Inc.
-# 
-# This file is part of GNU Radio
-# 
-# GNU Radio is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 3, or (at your option)
-# any later version.
-# 
-# GNU Radio is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-# 
-# You should have received a copy of the GNU General Public License
-# along with GNU Radio; see the file COPYING.  If not, write to
-# the Free Software Foundation, Inc., 51 Franklin Street,
-# Boston, MA 02110-1301, USA.
-# 
-
-# Makefile.swig.gen for audio_alsa.i
-
-## Default install locations for these files:
-##
-## Default location for the Python directory is:
-##  ${prefix}/lib/python${python_version}/site-packages/[category]/audio_alsa
-## Default location for the Python exec directory is:
-##  ${exec_prefix}/lib/python${python_version}/site-packages/[category]/audio_alsa
-##
-## The following can be overloaded to change the install location, but
-## this has to be done in the including Makefile.am -before-
-## Makefile.swig is included.
-
-audio_alsa_pythondir_category ?= gnuradio/audio_alsa
-audio_alsa_pylibdir_category ?= $(audio_alsa_pythondir_category)
-audio_alsa_pythondir = $(pythondir)/$(audio_alsa_pythondir_category)
-audio_alsa_pylibdir = $(pyexecdir)/$(audio_alsa_pylibdir_category)
-
-## SWIG headers are always installed into the same directory.
-
-audio_alsa_swigincludedir = $(swigincludedir)
-
-## This is a template file for a "generated" Makefile addition (in
-## this case, "Makefile.swig.gen").  By including the top-level
-## Makefile.swig, this file will be used to generate the SWIG
-## dependencies.  Assign the variable TOP_SWIG_FILES to be the list of
-## SWIG .i files to generated wrappings for; there can be more than 1
-## so long as the names are unique (no sorting is done on the
-## TOP_SWIG_FILES list). This file explicitly assumes that a SWIG .i
-## file will generate .cc, .py, and possibly .h files -- meaning that
-## all of these files will have the same base name (that provided for
-## the SWIG .i file).
-##
-## This code is setup to ensure parallel MAKE ("-j" or "-jN") does the
-## right thing.  For more info, see <
-## http://sources.redhat.com/automake/automake.html#Multiple-Outputs >
-
-## Stamps used to ensure parallel make does the right thing.  These
-## are removed by "make clean", but otherwise unused except during the
-## parallel built.  These will not be included in a tarball, because
-## the SWIG-generated files will be removed from the distribution.
-
-STAMPS += $(DEPDIR)/audio_alsa-generate-*
-
-## Other cleaned files: dependency files generated by SWIG or this Makefile
-
-MOSTLYCLEANFILES += $(DEPDIR)/*.S*
-
-## Add the .py and .cc files to the list of SWIG built sources.  The
-## .h file is sometimes built, but not always ... so that one has to
-## be added manually by the including Makefile.am .
-
-swig_built_sources += audio_alsa.py audio_alsa.cc
-
-## Various SWIG variables.  These can be overloaded in the including
-## Makefile.am by setting the variable value there, then including
-## Makefile.swig .
-
-audio_alsa_swiginclude_HEADERS =		\
-	audio_alsa.i			\
-	$(audio_alsa_swiginclude_headers)
-
-audio_alsa_pylib_LTLIBRARIES =		\
-	_audio_alsa.la
-
-_audio_alsa_la_SOURCES = 			\
-	audio_alsa.cc			\
-	$(audio_alsa_la_swig_sources)
-
-_audio_alsa_la_LIBADD =			\
-	$(STD_SWIG_LA_LIB_ADD)		\
-	$(audio_alsa_la_swig_libadd)
-
-_audio_alsa_la_LDFLAGS =			\
-	$(STD_SWIG_LA_LD_FLAGS)		\
-	$(audio_alsa_la_swig_ldflags)
-
-_audio_alsa_la_CXXFLAGS =			\
-	$(STD_SWIG_CXX_FLAGS)		\
-	$(audio_alsa_la_swig_cxxflags)
-
-audio_alsa_python_PYTHON =			\
-	audio_alsa.py			\
-	$(audio_alsa_python)
-
-## Entry rule for running SWIG
-
-audio_alsa.h audio_alsa.py audio_alsa.cc: audio_alsa.i
-## This rule will get called only when MAKE decides that one of the
-## targets needs to be created or re-created, because:
-##
-## * The .i file is newer than any or all of the generated files;
-##
-## * Any or all of the .cc, .h, or .py files does not exist and is
-##   needed (in the case this file is not needed, the rule for it is
-##   ignored); or
-##
-## * Some SWIG-based dependecy of the .cc file isn't met and hence the
-##   .cc file needs be be regenerated.  Explanation: Because MAKE
-##   knows how to handle dependencies for .cc files (regardless of
-##   their name or extension), then the .cc file is used as a target
-##   instead of the .i file -- but with the dependencies of the .i
-##   file.  It is this last reason why the line:
-##
-##     	if test -f $@; then :; else
-##
-##   cannot be used in this case: If a .i file dependecy is not met,
-##   then the .cc file needs to be rebuilt.  But if the stamp is newer
-##   than the .cc file, and the .cc file exists, then in the original
-##   version (with the 'test' above) the internal MAKE call will not
-##   be issued and hence the .cc file will not be rebuilt.
-##
-## Once execution gets to here, it should always proceed no matter the
-## state of a stamp (as discussed in link above).  The
-## $(DEPDIR)/audio_alsa-generate stuff is used to allow for parallel
-## builds to "do the right thing".  The stamp has no relationship with
-## either the target files or dependency file; it is used solely for
-## the protection of multiple builds during a given call to MAKE.
-##
-## Catch signals SIGHUP (1), SIGINT (2), SIGPIPE (13), and SIGTERM
-## (15).  At a caught signal, the quoted command will be issued before
-## exiting.  In this case, remove any stamp, whether temporary of not.
-## The trap is valid until the process exits; the process includes all
-## commands appended via "\"s.
-##
-	trap 'rm -rf $(DEPDIR)/audio_alsa-generate-*' 1 2 13 15; \
-##
-## Create a temporary directory, which acts as a lock.  The first
-## process to create the directory will succeed and issue the MAKE
-## command to do the actual work, while all subsequent processes will
-## fail -- leading them to wait for the first process to finish.
-##
-	if mkdir $(DEPDIR)/audio_alsa-generate-lock 2>/dev/null; then \
-##
-## This code is being executed by the first process to succeed in
-## creating the directory lock.
-##
-## Remove the stamp associated with this filename.
-##
-		rm -f $(DEPDIR)/audio_alsa-generate-stamp; \
-##
-## Tell MAKE to run the rule for creating this stamp.
-##
-		$(MAKE) $(AM_MAKEFLAGS) $(DEPDIR)/audio_alsa-generate-stamp WHAT=$<; \
-##
-## Now that the .cc, .h, and .py files have been (re)created from the
-## .i file, future checking of this rule during the same MAKE
-## execution will come back that the rule doesn't need to be executed
-## because none of the conditions mentioned at the start of this rule
-## will be positive.  Remove the the directory lock, which frees up
-## any waiting process(es) to continue.
-##
-		rmdir $(DEPDIR)/audio_alsa-generate-lock; \
-	else \
-##
-## This code is being executed by any follower processes while the
-## directory lock is in place.
-##
-## Wait until the first process is done, testing once per second.
-##
-		while test -d $(DEPDIR)/audio_alsa-generate-lock; do \
-			sleep 1; \
-		done; \
-##
-## Succeed if and only if the first process succeeded; exit this
-## process returning the status of the generated stamp.
-##
-		test -f $(DEPDIR)/audio_alsa-generate-stamp; \
-		exit $$?; \
-	fi;
-
-$(DEPDIR)/audio_alsa-generate-stamp:
-## This rule will be called only by the first process issuing the
-## above rule to succeed in creating the lock directory, after
-## removing the actual stamp file in order to guarantee that MAKE will
-## execute this rule.
-##
-## Call SWIG to generate the various output files; special
-## post-processing on 'mingw32' host OS for the dependency file.
-##
-	if $(SWIG) $(STD_SWIG_PYTHON_ARGS) $(audio_alsa_swig_args) \
-		-MD -MF $(DEPDIR)/audio_alsa.Std \
-		-module audio_alsa -o audio_alsa.cc $(WHAT); then \
-	    if test $(host_os) = mingw32; then \
-		$(RM) $(DEPDIR)/audio_alsa.Sd; \
-		$(SED) 's,\\\\,/,g' < $(DEPDIR)/audio_alsa.Std \
-			> $(DEPDIR)/audio_alsa.Sd; \
-		$(RM) $(DEPDIR)/audio_alsa.Std; \
-		$(MV) $(DEPDIR)/audio_alsa.Sd $(DEPDIR)/audio_alsa.Std; \
-	    fi; \
-	else \
-	    $(RM) $(DEPDIR)/audio_alsa.S*; exit 1; \
-	fi;
-##
-## Mess with the SWIG output .Std dependency file, to create a
-## dependecy file valid for the input .i file: Basically, simulate the
-## dependency file created for libraries by GNU's libtool for C++,
-## where all of the dependencies for the target are first listed, then
-## each individual dependency is listed as a target with no further
-## dependencies.
-##
-## (1) remove the current dependency file
-##
-	$(RM) $(DEPDIR)/audio_alsa.d
-##
-## (2) Copy the whole SWIG file:
-##
-	cp $(DEPDIR)/audio_alsa.Std $(DEPDIR)/audio_alsa.d
-##
-## (3) all a carriage return to the end of the dependency file.
-##
-	echo "" >> $(DEPDIR)/audio_alsa.d
-##
-## (4) from the SWIG file, remove the first line (the target); remove
-##     trailing " \" and " " from each line.  Append ":" to each line,
-##     followed by 2 carriage returns, then append this to the end of
-##     the dependency file.
-##
-	$(SED) -e '1d;s, \\,,g;s, ,,g' < $(DEPDIR)/audio_alsa.Std | \
-		awk '{ printf "%s:\n\n", $$0 }' >> $(DEPDIR)/audio_alsa.d
-##
-## (5) remove the SWIG-generated file
-##
-	$(RM) $(DEPDIR)/audio_alsa.Std
-##
-## Create the stamp for this filename generation, to signal success in
-## executing this rule; allows other threads waiting on this process
-## to continue.
-##
-	touch $(DEPDIR)/audio_alsa-generate-stamp
-
-# KLUDGE: Force runtime include of a SWIG dependency file.  This is
-# not guaranteed to be portable, but will probably work.  If it works,
-# we have accurate dependencies for our swig stuff, which is good.
-
-@am__include@ @am__quote@./$(DEPDIR)/audio_alsa.d@am__quote@
-
diff --git a/gr-audio-alsa/src/audio_alsa.i b/gr-audio-alsa/src/audio_alsa.i
deleted file mode 100644
index 88402333a..000000000
--- a/gr-audio-alsa/src/audio_alsa.i
+++ /dev/null
@@ -1,75 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004,2006,2009 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-%include "gnuradio.i"				// the common stuff
-
-%{
-#include "audio_alsa_sink.h"
-#include "audio_alsa_source.h"
-%}
-
-// ----------------------------------------------------------------
-
-GR_SWIG_BLOCK_MAGIC(audio_alsa,source)
-
-audio_alsa_source_sptr
-audio_alsa_make_source (int sampling_rate,
-			const std::string dev = "",
-			bool ok_to_block = true
-			) throw (std::runtime_error);
-
-class audio_alsa_source : public gr_sync_block {
-
- protected:
-  audio_alsa_source (int sampling_rate,
-		     const std::string device_name,
-		     bool ok_to_block = true
-		     ) throw (std::runtime_error);
-
- public:
-  ~audio_alsa_source ();
-
-  bool start();
-  bool stop();
-};
-
-// ----------------------------------------------------------------
-
-GR_SWIG_BLOCK_MAGIC(audio_alsa,sink)
-
-audio_alsa_sink_sptr
-audio_alsa_make_sink (int sampling_rate,
-		      const std::string dev = "",
-		      bool ok_to_block = true
-		      ) throw (std::runtime_error);
-
-class audio_alsa_sink : public gr_sync_block {
-
- protected:
-  audio_alsa_sink (int sampling_rate,
-		   const std::string device_name,
-		   bool ok_to_block
-		   ) throw (std::runtime_error);
-
- public:
-  ~audio_alsa_sink ();
-};
diff --git a/gr-audio-alsa/src/audio_alsa_sink.cc b/gr-audio-alsa/src/audio_alsa_sink.cc
deleted file mode 100644
index d44a93b3f..000000000
--- a/gr-audio-alsa/src/audio_alsa_sink.cc
+++ /dev/null
@@ -1,551 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004,2010 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <audio_alsa_sink.h>
-#include <gr_io_signature.h>
-#include <gr_prefs.h>
-#include <stdio.h>
-#include <iostream>
-#include <stdexcept>
-#include <gri_alsa.h>
-
-
-static bool CHATTY_DEBUG = false;
-
-
-static snd_pcm_format_t acceptable_formats[] = {
-  // these are in our preferred order...
-  SND_PCM_FORMAT_S32,
-  SND_PCM_FORMAT_S16
-};
-
-#define NELEMS(x) (sizeof(x)/sizeof(x[0]))
-
-
-static std::string 
-default_device_name ()
-{
-  return gr_prefs::singleton()->get_string("audio_alsa", "default_output_device", "hw:0,0");
-}
-
-static double
-default_period_time ()
-{
-  return std::max(0.001, gr_prefs::singleton()->get_double("audio_alsa", "period_time", 0.010));
-}
-
-static int
-default_nperiods ()
-{
-  return std::max(2L, gr_prefs::singleton()->get_long("audio_alsa", "nperiods", 4));
-}
-
-// ----------------------------------------------------------------
-
-audio_alsa_sink_sptr
-audio_alsa_make_sink (int sampling_rate,
-		      const std::string dev,
-		      bool ok_to_block)
-{
-  return gnuradio::get_initial_sptr(new audio_alsa_sink (sampling_rate, dev,
-						    ok_to_block));
-}
-
-audio_alsa_sink::audio_alsa_sink (int sampling_rate,
-				  const std::string device_name,
-				  bool ok_to_block)
-  : gr_sync_block ("audio_alsa_sink",
-		   gr_make_io_signature (0, 0, 0),
-		   gr_make_io_signature (0, 0, 0)),
-    d_sampling_rate (sampling_rate),
-    d_device_name (device_name.empty() ? default_device_name() : device_name),
-    d_pcm_handle (0),
-    d_hw_params ((snd_pcm_hw_params_t *)(new char[snd_pcm_hw_params_sizeof()])),
-    d_sw_params ((snd_pcm_sw_params_t *)(new char[snd_pcm_sw_params_sizeof()])),
-    d_nperiods (default_nperiods()),
-    d_period_time_us ((unsigned int) (default_period_time() * 1e6)),
-    d_period_size (0),
-    d_buffer_size_bytes (0), d_buffer (0),
-    d_worker (0), d_special_case_mono_to_stereo (false),
-    d_nunderuns (0), d_nsuspends (0), d_ok_to_block(ok_to_block)
-{
-  CHATTY_DEBUG = gr_prefs::singleton()->get_bool("audio_alsa", "verbose", false);
-
-  int  	error;
-  int	dir;
-
-  // open the device for playback
-  error = snd_pcm_open(&d_pcm_handle, d_device_name.c_str (),
-		       SND_PCM_STREAM_PLAYBACK, 0);
-  if (ok_to_block == false)
-    snd_pcm_nonblock(d_pcm_handle, !ok_to_block);
-  if (error < 0){
-    fprintf (stderr, "audio_alsa_sink[%s]: %s\n",
-	     d_device_name.c_str(), snd_strerror(error));
-    throw std::runtime_error ("audio_alsa_sink");
-  }
-
-  // Fill params with a full configuration space for a PCM.
-  error = snd_pcm_hw_params_any(d_pcm_handle, d_hw_params);
-  if (error < 0)
-    bail ("broken configuration for playback", error);
-
-
-  if (CHATTY_DEBUG)
-    gri_alsa_dump_hw_params (d_pcm_handle, d_hw_params, stdout);
-
-
-  // now that we know how many channels the h/w can handle, set input signature
-  unsigned int umin_chan, umax_chan;
-  snd_pcm_hw_params_get_channels_min (d_hw_params, &umin_chan);
-  snd_pcm_hw_params_get_channels_max (d_hw_params, &umax_chan);
-  int min_chan = std::min (umin_chan, 1000U);
-  int max_chan = std::min (umax_chan, 1000U);
-
-  // As a special case, if the hw's min_chan is two, we'll accept
-  // a single input and handle the duplication ourselves.
-
-  if (min_chan == 2){
-    min_chan = 1;
-    d_special_case_mono_to_stereo = true;
-  }
-  set_input_signature (gr_make_io_signature (min_chan, max_chan,
-					     sizeof (float)));
-  
-  // fill in portions of the d_hw_params that we know now...
-
-  // Specify the access methods we implement
-  // For now, we only handle RW_INTERLEAVED...
-  snd_pcm_access_mask_t *access_mask;
-  snd_pcm_access_mask_t **access_mask_ptr = &access_mask; // FIXME: workaround for compiler warning
-  snd_pcm_access_mask_alloca (access_mask_ptr);
-  snd_pcm_access_mask_none (access_mask);
-  snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_INTERLEAVED);
-  // snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_NONINTERLEAVED);
-
-  if ((error = snd_pcm_hw_params_set_access_mask (d_pcm_handle,
-						  d_hw_params, access_mask)) < 0)
-    bail ("failed to set access mask", error);
-
-
-  // set sample format
-  if (!gri_alsa_pick_acceptable_format (d_pcm_handle, d_hw_params,
-					acceptable_formats,
-					NELEMS (acceptable_formats),
-					&d_format,
-					"audio_alsa_sink",
-					CHATTY_DEBUG))
-    throw std::runtime_error ("audio_alsa_sink");
-  
-
-  // sampling rate
-  unsigned int orig_sampling_rate = d_sampling_rate;
-  if ((error = snd_pcm_hw_params_set_rate_near (d_pcm_handle, d_hw_params,
-						&d_sampling_rate, 0)) < 0)
-    bail ("failed to set rate near", error);
-  
-  if (orig_sampling_rate != d_sampling_rate){
-    fprintf (stderr, "audio_alsa_sink[%s]: unable to support sampling rate %d\n",
-	     snd_pcm_name (d_pcm_handle), orig_sampling_rate);
-    fprintf (stderr, "  card requested %d instead.\n", d_sampling_rate);
-  }
-
-  /*
-   * ALSA transfers data in units of "periods".
-   * We indirectly determine the underlying buffersize by specifying
-   * the number of periods we want (typically 4) and the length of each
-   * period in units of time (typically 1ms).
-   */
-  unsigned int min_nperiods, max_nperiods;
-  snd_pcm_hw_params_get_periods_min (d_hw_params, &min_nperiods, &dir);
-  snd_pcm_hw_params_get_periods_max (d_hw_params, &max_nperiods, &dir);
-  //fprintf (stderr, "alsa_sink: min_nperiods = %d, max_nperiods = %d\n",
-  // min_nperiods, max_nperiods);
-
-  unsigned int orig_nperiods = d_nperiods;
-  d_nperiods = std::min (std::max (min_nperiods, d_nperiods), max_nperiods);
-
-  // adjust period time so that total buffering remains more-or-less constant
-  d_period_time_us = (d_period_time_us * orig_nperiods) / d_nperiods;
-
-  error = snd_pcm_hw_params_set_periods (d_pcm_handle, d_hw_params,
-					 d_nperiods, 0);
-  if (error < 0)
-    bail ("set_periods failed", error);
-
-  dir = 0;
-  error = snd_pcm_hw_params_set_period_time_near (d_pcm_handle, d_hw_params,
-						  &d_period_time_us, &dir);
-  if (error < 0)
-    bail ("set_period_time_near failed", error);
-
-  dir = 0;
-  error = snd_pcm_hw_params_get_period_size (d_hw_params,
-					     &d_period_size, &dir);
-  if (error < 0)
-    bail ("get_period_size failed", error);
-  
-  set_output_multiple (d_period_size);
-}
-
-
-bool
-audio_alsa_sink::check_topology (int ninputs, int noutputs)
-{
-  // ninputs is how many channels the user has connected.
-  // Now we can finish up setting up the hw params...
-
-  int nchan = ninputs;
-  int err;
-
-  // Check the state of the stream
-  // Ensure that the pcm is in a state where we can still mess with the hw_params
-  snd_pcm_state_t state;
-  state=snd_pcm_state(d_pcm_handle);
-  if ( state== SND_PCM_STATE_RUNNING)
-    return true;  // If stream is running, don't change any parameters
-  else if(state == SND_PCM_STATE_XRUN )
-    snd_pcm_prepare ( d_pcm_handle ); // Prepare stream on underrun, and we can set parameters;
-  
-  bool special_case = nchan == 1 && d_special_case_mono_to_stereo;
-  if (special_case)
-    nchan = 2;
-  
-  err = snd_pcm_hw_params_set_channels (d_pcm_handle, d_hw_params, nchan);
-
-  if (err < 0){
-    output_error_msg ("set_channels failed", err);
-    return false;
-  }
-
-  // set the parameters into the driver...
-  err = snd_pcm_hw_params(d_pcm_handle, d_hw_params);
-  if (err < 0){
-    output_error_msg ("snd_pcm_hw_params failed", err);
-    return false;
-  }
-
-  // get current s/w params
-  err = snd_pcm_sw_params_current (d_pcm_handle, d_sw_params);
-  if (err < 0)
-    bail ("snd_pcm_sw_params_current", err);
-  
-  // Tell the PCM device to wait to start until we've filled
-  // it's buffers half way full.  This helps avoid audio underruns.
-
-  err = snd_pcm_sw_params_set_start_threshold(d_pcm_handle,
-					      d_sw_params,
-					      d_nperiods * d_period_size / 2);
-  if (err < 0)
-    bail ("snd_pcm_sw_params_set_start_threshold", err);
-
-  // store the s/w params
-  err = snd_pcm_sw_params (d_pcm_handle, d_sw_params);
-  if (err < 0)
-    bail ("snd_pcm_sw_params", err);
-
-  d_buffer_size_bytes =
-    d_period_size * nchan * snd_pcm_format_size (d_format, 1);
-
-  d_buffer = new char [d_buffer_size_bytes];
-
-  if (CHATTY_DEBUG)
-    fprintf (stdout, "audio_alsa_sink[%s]: sample resolution = %d bits\n",
-	     snd_pcm_name (d_pcm_handle),
-	     snd_pcm_hw_params_get_sbits (d_hw_params));
-
-  switch (d_format){
-  case SND_PCM_FORMAT_S16:
-    if (special_case)
-      d_worker = &audio_alsa_sink::work_s16_1x2;
-    else
-      d_worker = &audio_alsa_sink::work_s16;
-    break;
-
-  case SND_PCM_FORMAT_S32:
-    if (special_case)
-      d_worker = &audio_alsa_sink::work_s32_1x2;
-    else
-      d_worker = &audio_alsa_sink::work_s32;
-    break;
-
-  default:
-    assert (0);
-  }
-  return true;
-}
-
-audio_alsa_sink::~audio_alsa_sink ()
-{
-  if (snd_pcm_state (d_pcm_handle) == SND_PCM_STATE_RUNNING)
-    snd_pcm_drop (d_pcm_handle);
-
-  snd_pcm_close(d_pcm_handle);
-  delete [] ((char *) d_hw_params);
-  delete [] ((char *) d_sw_params);
-  delete [] d_buffer;
-}
-
-int
-audio_alsa_sink::work (int noutput_items,
-		       gr_vector_const_void_star &input_items,
-		       gr_vector_void_star &output_items)
-{
-  assert ((noutput_items % d_period_size) == 0);
-
-  // this is a call through a pointer to a method...
-  return (this->*d_worker)(noutput_items, input_items, output_items);
-}
-
-/*
- * Work function that deals with float to S16 conversion
- */
-int
-audio_alsa_sink::work_s16 (int noutput_items,
-			   gr_vector_const_void_star &input_items,
-			   gr_vector_void_star &output_items)
-{
-  typedef gr_int16	sample_t;	// the type of samples we're creating
-  static const int NBITS = 16;		// # of bits in a sample
-  
-  unsigned int nchan = input_items.size ();
-  const float **in = (const float **) &input_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-  int n;
-
-  unsigned int sizeof_frame = nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  for (n = 0; n < noutput_items; n += d_period_size){
-
-    // process one period of data
-    bi = 0;
-    for (unsigned int i = 0; i < d_period_size; i++){
-      for (unsigned int chan = 0; chan < nchan; chan++){
-	buf[bi++] = (sample_t) (in[chan][i] * (float) ((1L << (NBITS-1)) - 1));
-      }
-    }
-
-    // update src pointers
-    for (unsigned int chan = 0; chan < nchan; chan++)
-      in[chan] += d_period_size;
-
-    if (!write_buffer (buf, d_period_size, sizeof_frame))  
-      return -1;	// No fixing this problem.  Say we're done.
-  }
-
-  return n;
-}
-
-
-/*
- * Work function that deals with float to S32 conversion
- */
-int
-audio_alsa_sink::work_s32 (int noutput_items,
-			   gr_vector_const_void_star &input_items,
-			   gr_vector_void_star &output_items)
-{
-  typedef gr_int32	sample_t;	// the type of samples we're creating
-  static const int NBITS = 32;		// # of bits in a sample
-  
-  unsigned int nchan = input_items.size ();
-  const float **in = (const float **) &input_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-  int n;
-
-  unsigned int sizeof_frame = nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  for (n = 0; n < noutput_items; n += d_period_size){
-
-    // process one period of data
-    bi = 0;
-    for (unsigned int i = 0; i < d_period_size; i++){
-      for (unsigned int chan = 0; chan < nchan; chan++){
-	buf[bi++] = (sample_t) (in[chan][i] * (float) ((1L << (NBITS-1)) - 1));
-      }
-    }
-
-    // update src pointers
-    for (unsigned int chan = 0; chan < nchan; chan++)
-      in[chan] += d_period_size;
-
-    if (!write_buffer (buf, d_period_size, sizeof_frame))  
-      return -1;	// No fixing this problem.  Say we're done.
-  }
-
-  return n;
-}
-
-/*
- * Work function that deals with float to S16 conversion and
- * mono to stereo kludge.
- */
-int
-audio_alsa_sink::work_s16_1x2 (int noutput_items,
-			       gr_vector_const_void_star &input_items,
-			       gr_vector_void_star &output_items)
-{
-  typedef gr_int16	sample_t;	// the type of samples we're creating
-  static const int NBITS = 16;		// # of bits in a sample
-  
-  assert (input_items.size () == 1);
-  static const unsigned int nchan = 2;
-  const float **in = (const float **) &input_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-  int n;
-
-  unsigned int sizeof_frame = nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  for (n = 0; n < noutput_items; n += d_period_size){
-
-    // process one period of data
-    bi = 0;
-    for (unsigned int i = 0; i < d_period_size; i++){
-      sample_t t = (sample_t) (in[0][i] * (float) ((1L << (NBITS-1)) - 1));
-      buf[bi++] = t;
-      buf[bi++] = t;
-    }
-
-    // update src pointers
-    in[0] += d_period_size;
-
-    if (!write_buffer (buf, d_period_size, sizeof_frame))  
-      return -1;	// No fixing this problem.  Say we're done.
-  }
-
-  return n;
-}
-
-/*
- * Work function that deals with float to S32 conversion and
- * mono to stereo kludge.
- */
-int
-audio_alsa_sink::work_s32_1x2 (int noutput_items,
-			       gr_vector_const_void_star &input_items,
-			       gr_vector_void_star &output_items)
-{
-  typedef gr_int32	sample_t;	// the type of samples we're creating
-  static const int NBITS = 32;		// # of bits in a sample
-  
-  assert (input_items.size () == 1);
-  static unsigned int nchan = 2;
-  const float **in = (const float **) &input_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-  int n;
-
-  unsigned int sizeof_frame = nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  for (n = 0; n < noutput_items; n += d_period_size){
-
-    // process one period of data
-    bi = 0;
-    for (unsigned int i = 0; i < d_period_size; i++){
-      sample_t t = (sample_t) (in[0][i] * (float) ((1L << (NBITS-1)) - 1));
-      buf[bi++] = t;
-      buf[bi++] = t;
-    }
-
-    // update src pointers
-    in[0] += d_period_size;
-
-    if (!write_buffer (buf, d_period_size, sizeof_frame))  
-      return -1;	// No fixing this problem.  Say we're done.
-  }
-
-  return n;
-}
-
-bool
-audio_alsa_sink::write_buffer (const void *vbuffer,
-			       unsigned nframes, unsigned sizeof_frame)
-{
-  const unsigned char *buffer = (const unsigned char *) vbuffer;
-
-  while (nframes > 0){
-    int r = snd_pcm_writei (d_pcm_handle, buffer, nframes);
-    if (r == -EAGAIN)
-    {
-      if (d_ok_to_block == true)
-	continue;		// try again
-      
-      break;
-    }
-
-    else if (r == -EPIPE){	// underrun
-      d_nunderuns++;
-      fputs ("aU", stderr);
-      if ((r = snd_pcm_prepare (d_pcm_handle)) < 0){
-	output_error_msg ("snd_pcm_prepare failed. Can't recover from underrun", r);
-	return false;
-      }
-      continue;			// try again
-    }
-
-    else if (r == -ESTRPIPE){	// h/w is suspended (whatever that means)
-				// This is apparently related to power management
-      d_nsuspends++;
-      if ((r = snd_pcm_resume (d_pcm_handle)) < 0){
-	output_error_msg ("failed to resume from suspend", r);
-	return false;
-      }
-      continue;			// try again
-    }
-
-    else if (r < 0){
-      output_error_msg ("snd_pcm_writei failed", r);
-      return false;
-    }
-
-    nframes -= r;
-    buffer += r * sizeof_frame;
-  }
-
-  return true;
-}
-
-
-void
-audio_alsa_sink::output_error_msg (const char *msg, int err)
-{
-  fprintf (stderr, "audio_alsa_sink[%s]: %s: %s\n",
-	   snd_pcm_name (d_pcm_handle), msg,  snd_strerror (err));
-}
-
-void
-audio_alsa_sink::bail (const char *msg, int err) throw (std::runtime_error)
-{
-  output_error_msg (msg, err);
-  throw std::runtime_error ("audio_alsa_sink");
-}
diff --git a/gr-audio-alsa/src/audio_alsa_sink.h b/gr-audio-alsa/src/audio_alsa_sink.h
deleted file mode 100644
index f3007f60f..000000000
--- a/gr-audio-alsa/src/audio_alsa_sink.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef INCLUDED_AUDIO_ALSA_SINK_H
-#define INCLUDED_AUDIO_ALSA_SINK_H
-
-// use new ALSA API
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-#include <gr_sync_block.h>
-#include <string>
-#include <alsa/asoundlib.h>
-#include <stdexcept>
-
-
-class audio_alsa_sink;
-typedef boost::shared_ptr<audio_alsa_sink> audio_alsa_sink_sptr;
-
-/*!
- * \brief make an alsa audio sink.
- *
- * \param sampling_rate	sampling rate in Hz
- * \param device_name ALSA pcm device name, e.g., "hw:0,0"
- * \param ok_to_block (currently ignored)
- */
-audio_alsa_sink_sptr
-audio_alsa_make_sink (int sampling_rate,
-		      const std::string device_name = "",
-		      bool ok_to_block = true);
-
-/*!
- * \brief audio sink using ALSA
- *
- * The sink has N input streams of floats, where N depends
- * on the hardware characteristics of the selected device.
- *
- * Input samples must be in the range [-1,1].
- */
-class audio_alsa_sink : public gr_sync_block {
-  friend audio_alsa_sink_sptr
-  audio_alsa_make_sink (int sampling_rate, const std::string device_name,
-			bool ok_to_block);
-
-  // typedef for pointer to class work method
-  typedef int (audio_alsa_sink::*work_t)(int noutput_items,
-					 gr_vector_const_void_star &input_items,
-					 gr_vector_void_star &output_items);
-
-  unsigned int		d_sampling_rate;
-  std::string		d_device_name;
-  snd_pcm_t            *d_pcm_handle;
-  snd_pcm_hw_params_t  *d_hw_params;
-  snd_pcm_sw_params_t  *d_sw_params;
-  snd_pcm_format_t	d_format;
-  unsigned int		d_nperiods;
-  unsigned int		d_period_time_us;	// microseconds
-  snd_pcm_uframes_t	d_period_size;		// in frames
-  unsigned int		d_buffer_size_bytes;	// sizeof of d_buffer
-  char		       *d_buffer;
-  work_t	        d_worker;		// the work method to use
-  bool			d_special_case_mono_to_stereo;
-
-  // random stats
-  int			d_nunderuns;		// count of underruns
-  int			d_nsuspends;		// count of suspends
-  bool			d_ok_to_block;      // defaults to "true", controls blocking/non-block I/O
-
-  void output_error_msg (const char *msg, int err);
-  void bail (const char *msg, int err) throw (std::runtime_error);
-
- protected:
-  audio_alsa_sink (int sampling_rate, const std::string device_name,
-		   bool ok_to_block);
-
- public:
-  ~audio_alsa_sink ();
-  
-  bool check_topology (int ninputs, int noutputs);
-
-  int work (int noutput_items,
-	    gr_vector_const_void_star &input_items,
-	    gr_vector_void_star &output_items);
-
-
-protected:
-  bool write_buffer (const void *buffer, unsigned nframes, unsigned sizeof_frame);
-
-  int work_s16 (int noutput_items,
-		gr_vector_const_void_star &input_items,
-		gr_vector_void_star &output_items);
-
-  int work_s16_1x2 (int noutput_items,
-		    gr_vector_const_void_star &input_items,
-		    gr_vector_void_star &output_items);
-
-  int work_s32 (int noutput_items,
-		gr_vector_const_void_star &input_items,
-		gr_vector_void_star &output_items);
-
-  int work_s32_1x2 (int noutput_items,
-		    gr_vector_const_void_star &input_items,
-		    gr_vector_void_star &output_items);
-};
-
-#endif /* INCLUDED_AUDIO_ALSA_SINK_H */
diff --git a/gr-audio-alsa/src/audio_alsa_source.cc b/gr-audio-alsa/src/audio_alsa_source.cc
deleted file mode 100644
index 7b3d91b9d..000000000
--- a/gr-audio-alsa/src/audio_alsa_source.cc
+++ /dev/null
@@ -1,507 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004,2006,2010 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <audio_alsa_source.h>
-#include <gr_io_signature.h>
-#include <gr_prefs.h>
-#include <stdio.h>
-#include <iostream>
-#include <stdexcept>
-#include <gri_alsa.h>
-
-
-static bool CHATTY_DEBUG = false;
-
-static snd_pcm_format_t acceptable_formats[] = {
-  // these are in our preferred order...
-  SND_PCM_FORMAT_S32,
-  SND_PCM_FORMAT_S16
-};
-
-#define NELEMS(x) (sizeof(x)/sizeof(x[0]))
-
-
-static std::string 
-default_device_name ()
-{
-  return gr_prefs::singleton()->get_string("audio_alsa", "default_input_device", "hw:0,0");
-}
-
-static double
-default_period_time ()
-{
-  return std::max(0.001, gr_prefs::singleton()->get_double("audio_alsa", "period_time", 0.010));
-}
-
-static int
-default_nperiods ()
-{
-  return std::max(2L, gr_prefs::singleton()->get_long("audio_alsa", "nperiods", 4));
-}
-
-// ----------------------------------------------------------------
-
-audio_alsa_source_sptr
-audio_alsa_make_source (int sampling_rate, const std::string dev,
-			bool ok_to_block)
-{
-  return gnuradio::get_initial_sptr(new audio_alsa_source (sampling_rate, dev,
-							ok_to_block));
-}
-
-audio_alsa_source::audio_alsa_source (int sampling_rate,
-				      const std::string device_name,
-				      bool ok_to_block)
-  : gr_sync_block ("audio_alsa_source",
-		   gr_make_io_signature (0, 0, 0),
-		   gr_make_io_signature (0, 0, 0)),
-    d_sampling_rate (sampling_rate),
-    d_device_name (device_name.empty() ? default_device_name() : device_name),
-    d_pcm_handle (0),
-    d_hw_params ((snd_pcm_hw_params_t *)(new char[snd_pcm_hw_params_sizeof()])),
-    d_sw_params ((snd_pcm_sw_params_t *)(new char[snd_pcm_sw_params_sizeof()])),
-    d_nperiods (default_nperiods()),
-    d_period_time_us ((unsigned int) (default_period_time() * 1e6)),
-    d_period_size (0),
-    d_buffer_size_bytes (0), d_buffer (0),
-    d_worker (0), d_hw_nchan (0),
-    d_special_case_stereo_to_mono (false),
-    d_noverruns (0), d_nsuspends (0)
-{
-
-  CHATTY_DEBUG = gr_prefs::singleton()->get_bool("audio_alsa", "verbose", false);
-
-  int  	error;
-  int	dir;
-
-  // open the device for capture
-  error = snd_pcm_open(&d_pcm_handle, d_device_name.c_str (),
-		       SND_PCM_STREAM_CAPTURE, 0);
-  if (error < 0){
-    fprintf (stderr, "audio_alsa_source[%s]: %s\n",
-	     d_device_name.c_str(), snd_strerror(error));
-    throw std::runtime_error ("audio_alsa_source");
-  }
-
-  // Fill params with a full configuration space for a PCM.
-  error = snd_pcm_hw_params_any(d_pcm_handle, d_hw_params);
-  if (error < 0)
-    bail ("broken configuration for playback", error);
-
-  if (CHATTY_DEBUG)
-    gri_alsa_dump_hw_params (d_pcm_handle, d_hw_params, stdout);
-
-  // now that we know how many channels the h/w can handle, set output signature
-  unsigned int umax_chan;
-  unsigned int umin_chan;
-  snd_pcm_hw_params_get_channels_min (d_hw_params, &umin_chan);
-  snd_pcm_hw_params_get_channels_max (d_hw_params, &umax_chan);
-  int min_chan = std::min (umin_chan, 1000U);
-  int max_chan = std::min (umax_chan, 1000U);
-
-  // As a special case, if the hw's min_chan is two, we'll accept
-  // a single output and handle the demux ourselves.
-
-  if (min_chan == 2){
-    min_chan = 1;
-    d_special_case_stereo_to_mono = true;
-  }
-  
-  set_output_signature (gr_make_io_signature (min_chan, max_chan,
-					      sizeof (float)));
-  
-  // fill in portions of the d_hw_params that we know now...
-
-  // Specify the access methods we implement
-  // For now, we only handle RW_INTERLEAVED...
-  snd_pcm_access_mask_t *access_mask;
-  snd_pcm_access_mask_t **access_mask_ptr = &access_mask; // FIXME: workaround for compiler warning
-  snd_pcm_access_mask_alloca (access_mask_ptr);
-  snd_pcm_access_mask_none (access_mask);
-  snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_INTERLEAVED);
-  // snd_pcm_access_mask_set (access_mask, SND_PCM_ACCESS_RW_NONINTERLEAVED);
-
-  if ((error = snd_pcm_hw_params_set_access_mask (d_pcm_handle,
-						  d_hw_params, access_mask)) < 0)
-    bail ("failed to set access mask", error);
-
-
-  // set sample format
-  if (!gri_alsa_pick_acceptable_format (d_pcm_handle, d_hw_params,
-					acceptable_formats,
-					NELEMS (acceptable_formats),
-					&d_format,
-					"audio_alsa_source",
-					CHATTY_DEBUG))
-    throw std::runtime_error ("audio_alsa_source");
-  
-
-  // sampling rate
-  unsigned int orig_sampling_rate = d_sampling_rate;
-  if ((error = snd_pcm_hw_params_set_rate_near (d_pcm_handle, d_hw_params,
-						&d_sampling_rate, 0)) < 0)
-    bail ("failed to set rate near", error);
-  
-  if (orig_sampling_rate != d_sampling_rate){
-    fprintf (stderr, "audio_alsa_source[%s]: unable to support sampling rate %d\n",
-	     snd_pcm_name (d_pcm_handle), orig_sampling_rate);
-    fprintf (stderr, "  card requested %d instead.\n", d_sampling_rate);
-  }
-
-  /*
-   * ALSA transfers data in units of "periods".
-   * We indirectly determine the underlying buffersize by specifying
-   * the number of periods we want (typically 4) and the length of each
-   * period in units of time (typically 1ms).
-   */
-  unsigned int min_nperiods, max_nperiods;
-  snd_pcm_hw_params_get_periods_min (d_hw_params, &min_nperiods, &dir);
-  snd_pcm_hw_params_get_periods_max (d_hw_params, &max_nperiods, &dir);
-  //fprintf (stderr, "alsa_source: min_nperiods = %d, max_nperiods = %d\n",
-  // min_nperiods, max_nperiods);
-
-
-  unsigned int orig_nperiods = d_nperiods;
-  d_nperiods = std::min (std::max (min_nperiods, d_nperiods), max_nperiods);
-
-  // adjust period time so that total buffering remains more-or-less constant
-  d_period_time_us = (d_period_time_us * orig_nperiods) / d_nperiods;
-
-  error = snd_pcm_hw_params_set_periods (d_pcm_handle, d_hw_params,
-					 d_nperiods, 0);
-  if (error < 0)
-    bail ("set_periods failed", error);
-
-  dir = 0;
-  error = snd_pcm_hw_params_set_period_time_near (d_pcm_handle, d_hw_params,
-						  &d_period_time_us, &dir);
-  if (error < 0)
-    bail ("set_period_time_near failed", error);
-
-  dir = 0;
-  error = snd_pcm_hw_params_get_period_size (d_hw_params,
-					     &d_period_size, &dir);
-  if (error < 0)
-    bail ("get_period_size failed", error);
-  
-  set_output_multiple (d_period_size);
-}
-
-bool
-audio_alsa_source::check_topology (int ninputs, int noutputs)
-{
-  // noutputs is how many channels the user has connected.
-  // Now we can finish up setting up the hw params...
-
-  unsigned int nchan = noutputs;
-  int err;
-
-  // FIXME check_topology may be called more than once.
-  // Ensure that the pcm is in a state where we can still mess with the hw_params
-
-  bool special_case = nchan == 1 && d_special_case_stereo_to_mono;
-  if (special_case)
-    nchan = 2;
-
-  d_hw_nchan = nchan;
-  err = snd_pcm_hw_params_set_channels (d_pcm_handle, d_hw_params, d_hw_nchan);
-  if (err < 0){
-    output_error_msg ("set_channels failed", err);
-    return false;
-  }
-
-  // set the parameters into the driver...
-  err = snd_pcm_hw_params(d_pcm_handle, d_hw_params);
-  if (err < 0){
-    output_error_msg ("snd_pcm_hw_params failed", err);
-    return false;
-  }
-
-  d_buffer_size_bytes =
-    d_period_size * d_hw_nchan * snd_pcm_format_size (d_format, 1);
-
-  d_buffer = new char [d_buffer_size_bytes];
-
-  if (CHATTY_DEBUG)
-    fprintf (stdout, "audio_alsa_source[%s]: sample resolution = %d bits\n",
-	     snd_pcm_name (d_pcm_handle),
-	     snd_pcm_hw_params_get_sbits (d_hw_params));
-
-  switch (d_format){
-  case SND_PCM_FORMAT_S16:
-    if (special_case)
-      d_worker = &audio_alsa_source::work_s16_2x1;
-    else
-      d_worker = &audio_alsa_source::work_s16;
-    break;
-
-  case SND_PCM_FORMAT_S32:
-    if (special_case)
-      d_worker = &audio_alsa_source::work_s32_2x1;
-    else
-      d_worker = &audio_alsa_source::work_s32;
-    break;
-
-  default:
-    assert (0);
-  }
-
-  return true;
-}
-
-audio_alsa_source::~audio_alsa_source ()
-{
-  if (snd_pcm_state (d_pcm_handle) == SND_PCM_STATE_RUNNING)
-    snd_pcm_drop (d_pcm_handle);
-
-  snd_pcm_close(d_pcm_handle);
-  delete [] ((char *) d_hw_params);
-  delete [] ((char *) d_sw_params);
-  delete [] d_buffer;
-}
-
-int
-audio_alsa_source::work (int noutput_items,
-			 gr_vector_const_void_star &input_items,
-			 gr_vector_void_star &output_items)
-{
-  assert ((noutput_items % d_period_size) == 0);
-  assert (noutput_items != 0);
-
-  // this is a call through a pointer to a method...
-  return (this->*d_worker)(noutput_items, input_items, output_items);
-}
-
-/*
- * Work function that deals with float to S16 conversion
- */
-int
-audio_alsa_source::work_s16 (int noutput_items,
-			     gr_vector_const_void_star &input_items,
-			     gr_vector_void_star &output_items)
-{
-  typedef gr_int16	sample_t;	// the type of samples we're creating
-  static const int NBITS = 16;		// # of bits in a sample
-  
-  unsigned int nchan = output_items.size ();
-  float **out = (float **) &output_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-
-  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  // To minimize latency, return at most a single period's worth of samples.
-  // [We could also read the first one in a blocking mode and subsequent
-  //  ones in non-blocking mode, but we'll leave that for later (or never).]
-  
-  if (!read_buffer (buf, d_period_size, sizeof_frame))
-    return -1;		// No fixing this problem.  Say we're done.
-
-  // process one period of data
-  bi = 0;
-  for (unsigned int i = 0; i < d_period_size; i++){
-    for (unsigned int chan = 0; chan < nchan; chan++){
-      out[chan][i] = (float) buf[bi++] * (1.0 / (float) ((1L << (NBITS-1)) - 1));
-    }
-  }
-
-  return d_period_size;
-}
-
-/*
- * Work function that deals with float to S16 conversion
- * and stereo to mono kludge...
- */
-int
-audio_alsa_source::work_s16_2x1 (int noutput_items,
-				 gr_vector_const_void_star &input_items,
-				 gr_vector_void_star &output_items)
-{
-  typedef gr_int16	sample_t;	// the type of samples we're creating
-  static const int NBITS = 16;		// # of bits in a sample
-  
-  unsigned int nchan = output_items.size ();
-  float **out = (float **) &output_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-
-  assert (nchan == 1);
-
-  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  // To minimize latency, return at most a single period's worth of samples.
-  // [We could also read the first one in a blocking mode and subsequent
-  //  ones in non-blocking mode, but we'll leave that for later (or never).]
-  
-  if (!read_buffer (buf, d_period_size, sizeof_frame))
-    return -1;		// No fixing this problem.  Say we're done.
-
-  // process one period of data
-  bi = 0;
-  for (unsigned int i = 0; i < d_period_size; i++){
-    int t = (buf[bi] + buf[bi+1]) / 2;
-    bi += 2;
-    out[0][i] = (float) t * (1.0 / (float) ((1L << (NBITS-1)) - 1));
-  }
-
-  return d_period_size;
-}
-
-/*
- * Work function that deals with float to S32 conversion
- */
-int
-audio_alsa_source::work_s32 (int noutput_items,
-			     gr_vector_const_void_star &input_items,
-			     gr_vector_void_star &output_items)
-{
-  typedef gr_int32	sample_t;	// the type of samples we're creating
-  static const int NBITS = 32;		// # of bits in a sample
-  
-  unsigned int nchan = output_items.size ();
-  float **out = (float **) &output_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-
-  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  // To minimize latency, return at most a single period's worth of samples.
-  // [We could also read the first one in a blocking mode and subsequent
-  //  ones in non-blocking mode, but we'll leave that for later (or never).]
-  
-  if (!read_buffer (buf, d_period_size, sizeof_frame))
-    return -1;		// No fixing this problem.  Say we're done.
-
-  // process one period of data
-  bi = 0;
-  for (unsigned int i = 0; i < d_period_size; i++){
-    for (unsigned int chan = 0; chan < nchan; chan++){
-      out[chan][i] = (float) buf[bi++] * (1.0 / (float) ((1L << (NBITS-1)) - 1));
-    }
-  }
-
-  return d_period_size;
-}
-
-/*
- * Work function that deals with float to S32 conversion
- * and stereo to mono kludge...
- */
-int
-audio_alsa_source::work_s32_2x1 (int noutput_items,
-				 gr_vector_const_void_star &input_items,
-				 gr_vector_void_star &output_items)
-{
-  typedef gr_int32	sample_t;	// the type of samples we're creating
-  static const int NBITS = 32;		// # of bits in a sample
-  
-  unsigned int nchan = output_items.size ();
-  float **out = (float **) &output_items[0];
-  sample_t *buf = (sample_t *) d_buffer;
-  int bi;
-
-  assert (nchan == 1);
-
-  unsigned int sizeof_frame = d_hw_nchan * sizeof (sample_t);
-  assert (d_buffer_size_bytes == d_period_size * sizeof_frame);
-
-  // To minimize latency, return at most a single period's worth of samples.
-  // [We could also read the first one in a blocking mode and subsequent
-  //  ones in non-blocking mode, but we'll leave that for later (or never).]
-  
-  if (!read_buffer (buf, d_period_size, sizeof_frame))
-    return -1;		// No fixing this problem.  Say we're done.
-
-  // process one period of data
-  bi = 0;
-  for (unsigned int i = 0; i < d_period_size; i++){
-    int t = (buf[bi] + buf[bi+1]) / 2;
-    bi += 2;
-    out[0][i] = (float) t * (1.0 / (float) ((1L << (NBITS-1)) - 1));
-  }
-
-  return d_period_size;
-}
-
-bool
-audio_alsa_source::read_buffer (void *vbuffer, unsigned nframes, unsigned sizeof_frame)
-{
-  unsigned char *buffer = (unsigned char *) vbuffer;
-
-  while (nframes > 0){
-    int r = snd_pcm_readi (d_pcm_handle, buffer, nframes);
-    if (r == -EAGAIN)
-      continue;			// try again
-
-    else if (r == -EPIPE){	// overrun
-      d_noverruns++;
-      fputs ("aO", stderr);
-      if ((r = snd_pcm_prepare (d_pcm_handle)) < 0){
-	output_error_msg ("snd_pcm_prepare failed. Can't recover from overrun", r);
-	return false;
-      }
-      continue;			// try again
-    }
-
-    else if (r == -ESTRPIPE){	// h/w is suspended (whatever that means)
-				// This is apparently related to power management
-      d_nsuspends++;
-      if ((r = snd_pcm_resume (d_pcm_handle)) < 0){
-	output_error_msg ("failed to resume from suspend", r);
-	return false;
-      }
-      continue;			// try again
-    }
-
-    else if (r < 0){
-      output_error_msg ("snd_pcm_readi failed", r);
-      return false;
-    }
-
-    nframes -= r;
-    buffer += r * sizeof_frame;
-  }
-
-  return true;
-}
-
-
-void
-audio_alsa_source::output_error_msg (const char *msg, int err)
-{
-  fprintf (stderr, "audio_alsa_source[%s]: %s: %s\n",
-	   snd_pcm_name (d_pcm_handle), msg,  snd_strerror (err));
-}
-
-void
-audio_alsa_source::bail (const char *msg, int err) throw (std::runtime_error)
-{
-  output_error_msg (msg, err);
-  throw std::runtime_error ("audio_alsa_source");
-}
diff --git a/gr-audio-alsa/src/audio_alsa_source.h b/gr-audio-alsa/src/audio_alsa_source.h
deleted file mode 100644
index db6c45683..000000000
--- a/gr-audio-alsa/src/audio_alsa_source.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004,2006 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef INCLUDED_AUDIO_ALSA_SOURCE_H
-#define INCLUDED_AUDIO_ALSA_SOURCE_H
-
-// use new ALSA API
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-#include <gr_sync_block.h>
-#include <string>
-#include <alsa/asoundlib.h>
-#include <stdexcept>
-
-class audio_alsa_source;
-typedef boost::shared_ptr<audio_alsa_source> audio_alsa_source_sptr;
-
-/*!
- * \brief Make an ALSA audio source.
- *
- * \param sampling_rate	sampling rate
- * \param device_name ALSA pcm device name, e.g., "hw:0,0"
- * \param ok_to_block  (currently ignored)
- */
-audio_alsa_source_sptr
-audio_alsa_make_source (int sampling_rate,
-			const std::string device_name = "",
-			bool ok_to_block = true);
-
-/*!
- * \brief audio source using ALSA
- *
- * The source has between 1 and N input streams of floats, where N is
- * depends on the hardware characteristics of the selected device.
- *
- * Output samples will be in the range [-1,1].
- */
-class audio_alsa_source : public gr_sync_block {
-  friend audio_alsa_source_sptr
-  audio_alsa_make_source (int sampling_rate,
-			  const std::string device_name,
-			  bool ok_to_block);
-
-  // typedef for pointer to class work method
-  typedef int (audio_alsa_source::*work_t)(int noutput_items,
-					   gr_vector_const_void_star &input_items,
-					   gr_vector_void_star &output_items);
-
-  unsigned int		d_sampling_rate;
-  std::string		d_device_name;
-  snd_pcm_t            *d_pcm_handle;
-  snd_pcm_hw_params_t  *d_hw_params;
-  snd_pcm_sw_params_t  *d_sw_params;
-  snd_pcm_format_t	d_format;
-  unsigned int		d_nperiods;
-  unsigned int		d_period_time_us;	// microseconds
-  snd_pcm_uframes_t	d_period_size;		// in frames
-  unsigned int		d_buffer_size_bytes;	// sizeof of d_buffer
-  char		       *d_buffer;
-  work_t	        d_worker;		// the work method to use
-  unsigned int		d_hw_nchan;		// # of configured h/w channels
-  bool			d_special_case_stereo_to_mono;
-
-  // random stats
-  int			d_noverruns;		// count of overruns
-  int			d_nsuspends;		// count of suspends
-
-  void output_error_msg (const char *msg, int err);
-  void bail (const char *msg, int err) throw (std::runtime_error);
-
- protected:
-  audio_alsa_source (int sampling_rate, const std::string device_name,
-		     bool ok_to_block);
-
- public:
-  ~audio_alsa_source ();
-  
-  bool check_topology (int ninputs, int noutputs);
-
-  int work (int noutput_items,
-	    gr_vector_const_void_star &input_items,
-	    gr_vector_void_star &output_items);
-
-protected:
-  bool read_buffer (void *buffer, unsigned nframes, unsigned sizeof_frame);
-
-  int work_s16 (int noutput_items,
-		gr_vector_const_void_star &input_items,
-		gr_vector_void_star &output_items);
-
-  int work_s16_2x1 (int noutput_items,
-		    gr_vector_const_void_star &input_items,
-		    gr_vector_void_star &output_items);
-
-  int work_s32 (int noutput_items,
-		gr_vector_const_void_star &input_items,
-		gr_vector_void_star &output_items);
-
-  int work_s32_2x1 (int noutput_items,
-		    gr_vector_const_void_star &input_items,
-		    gr_vector_void_star &output_items);
-};
-
-#endif /* INCLUDED_AUDIO_ALSA_SOURCE_H */
diff --git a/gr-audio-alsa/src/gri_alsa.cc b/gr-audio-alsa/src/gri_alsa.cc
deleted file mode 100644
index d9fda0f7d..000000000
--- a/gr-audio-alsa/src/gri_alsa.cc
+++ /dev/null
@@ -1,175 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <gri_alsa.h>
-#include <algorithm>
-
-static snd_pcm_access_t access_types[] = {
-  SND_PCM_ACCESS_MMAP_INTERLEAVED,
-  SND_PCM_ACCESS_MMAP_NONINTERLEAVED,
-  SND_PCM_ACCESS_MMAP_COMPLEX,
-  SND_PCM_ACCESS_RW_INTERLEAVED,
-  SND_PCM_ACCESS_RW_NONINTERLEAVED
-};
-
-static snd_pcm_format_t format_types[] = {
-  // SND_PCM_FORMAT_UNKNOWN,
-  SND_PCM_FORMAT_S8,
-  SND_PCM_FORMAT_U8,
-  SND_PCM_FORMAT_S16_LE,
-  SND_PCM_FORMAT_S16_BE,
-  SND_PCM_FORMAT_U16_LE,
-  SND_PCM_FORMAT_U16_BE,
-  SND_PCM_FORMAT_S24_LE,
-  SND_PCM_FORMAT_S24_BE,
-  SND_PCM_FORMAT_U24_LE,
-  SND_PCM_FORMAT_U24_BE,
-  SND_PCM_FORMAT_S32_LE,
-  SND_PCM_FORMAT_S32_BE,
-  SND_PCM_FORMAT_U32_LE,
-  SND_PCM_FORMAT_U32_BE,
-  SND_PCM_FORMAT_FLOAT_LE,
-  SND_PCM_FORMAT_FLOAT_BE,
-  SND_PCM_FORMAT_FLOAT64_LE,
-  SND_PCM_FORMAT_FLOAT64_BE,
-  SND_PCM_FORMAT_IEC958_SUBFRAME_LE,
-  SND_PCM_FORMAT_IEC958_SUBFRAME_BE,
-  SND_PCM_FORMAT_MU_LAW,
-  SND_PCM_FORMAT_A_LAW,
-  SND_PCM_FORMAT_IMA_ADPCM,
-  SND_PCM_FORMAT_MPEG,
-  SND_PCM_FORMAT_GSM,
-  SND_PCM_FORMAT_SPECIAL,
-  SND_PCM_FORMAT_S24_3LE,
-  SND_PCM_FORMAT_S24_3BE,
-  SND_PCM_FORMAT_U24_3LE,
-  SND_PCM_FORMAT_U24_3BE,
-  SND_PCM_FORMAT_S20_3LE,
-  SND_PCM_FORMAT_S20_3BE,
-  SND_PCM_FORMAT_U20_3LE,
-  SND_PCM_FORMAT_U20_3BE,
-  SND_PCM_FORMAT_S18_3LE,
-  SND_PCM_FORMAT_S18_3BE,
-  SND_PCM_FORMAT_U18_3LE,
-  SND_PCM_FORMAT_U18_3BE
-};
-
-static unsigned int test_rates[] = {
-  8000, 16000, 22050, 32000, 44100, 48000, 96000, 192000
-};
-
-#define NELEMS(x) (sizeof(x)/sizeof(x[0]))
-
-void
-gri_alsa_dump_hw_params (snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, FILE *fp)
-{
-  fprintf (fp, "PCM name: %s\n", snd_pcm_name (pcm));
-  
-  fprintf (fp, "Access types:\n");
-  for (unsigned i = 0; i < NELEMS (access_types); i++){
-    snd_pcm_access_t	at = access_types[i];
-    fprintf (fp, "    %-20s %s\n",
-	     snd_pcm_access_name (at),
-	     snd_pcm_hw_params_test_access (pcm, hwparams, at) == 0 ? "YES" : "NO");
-  }
-
-  fprintf (fp, "Formats:\n");
-  for (unsigned i = 0; i < NELEMS (format_types); i++){
-    snd_pcm_format_t	ft = format_types[i];
-    if (0)
-      fprintf (fp, "    %-20s %s\n",
-	       snd_pcm_format_name (ft),
-	       snd_pcm_hw_params_test_format (pcm, hwparams, ft) == 0 ? "YES" : "NO");
-    else {
-      if (snd_pcm_hw_params_test_format (pcm, hwparams, ft) == 0)
-	fprintf (fp, "    %-20s YES\n", snd_pcm_format_name (ft));
-    }
-  }
-
-  fprintf (fp, "Number of channels\n");
-  unsigned int min_chan, max_chan;
-  snd_pcm_hw_params_get_channels_min (hwparams, &min_chan);
-  snd_pcm_hw_params_get_channels_max (hwparams, &max_chan);
-  fprintf (fp, "    min channels: %d\n", min_chan);
-  fprintf (fp, "    max channels: %d\n", max_chan);
-  unsigned int chan;
-  max_chan = std::min (max_chan, 16U);	// truncate display...
-  for (chan = min_chan; chan <= max_chan; chan++){
-    fprintf (fp, "    %d channels\t%s\n", chan,
-	     snd_pcm_hw_params_test_channels (pcm, hwparams, chan) == 0 ? "YES" : "NO");
-  }
-
-  fprintf (fp, "Sample Rates:\n");
-  unsigned int min_rate, max_rate;
-  int	min_dir, max_dir;
-
-  snd_pcm_hw_params_get_rate_min (hwparams, &min_rate, &min_dir);
-  snd_pcm_hw_params_get_rate_max (hwparams, &max_rate, &max_dir);
-  fprintf (fp, "    min rate: %7d (dir = %d)\n", min_rate, min_dir);
-  fprintf (fp, "    max rate: %7d (dir = %d)\n", max_rate, max_dir);
-  for (unsigned i = 0; i < NELEMS (test_rates); i++){
-    unsigned int rate = test_rates[i];
-    fprintf (fp, "    %6u  %s\n", rate,
-	     snd_pcm_hw_params_test_rate (pcm, hwparams, rate, 0) == 0 ? "YES" : "NO");
-  }
-
-  fflush (fp);
-}
-
-bool
-gri_alsa_pick_acceptable_format (snd_pcm_t *pcm,
-				 snd_pcm_hw_params_t *hwparams,
-				 snd_pcm_format_t acceptable_formats[],
-				 unsigned nacceptable_formats,
-				 snd_pcm_format_t *selected_format,
-				 const char *error_msg_tag,
-				 bool verbose)
-{
-  int err;
-
-  // pick a format that we like...
-  for (unsigned i = 0; i < nacceptable_formats; i++){
-    if (snd_pcm_hw_params_test_format (pcm, hwparams,
-				       acceptable_formats[i]) == 0){
-      err = snd_pcm_hw_params_set_format (pcm, hwparams, acceptable_formats[i]);
-      if (err < 0){
-	fprintf (stderr, "%s[%s]: failed to set format: %s\n",
-		 error_msg_tag, snd_pcm_name (pcm), snd_strerror (err));
-	return false;
-      }
-      if (verbose)
-	fprintf (stdout, "%s[%s]: using %s\n",
-		 error_msg_tag, snd_pcm_name (pcm),
-		 snd_pcm_format_name (acceptable_formats[i]));
-      *selected_format = acceptable_formats[i];
-      return true;
-    }
-  }
-  
-  fprintf (stderr, "%s[%s]: failed to find acceptable format",
-	   error_msg_tag, snd_pcm_name (pcm));
-  return false;
-}
diff --git a/gr-audio-alsa/src/gri_alsa.h b/gr-audio-alsa/src/gri_alsa.h
deleted file mode 100644
index 3d72fd950..000000000
--- a/gr-audio-alsa/src/gri_alsa.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2004 Free Software Foundation, Inc.
- * 
- * This file is part of GNU Radio
- * 
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- * 
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING.  If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef INCLUDED_GRI_ALSA_H
-#define INCLUDED_GRI_ALSA_H
-
-#include <stdio.h>
-#include <alsa/asoundlib.h>
-
-void 
-gri_alsa_dump_hw_params (snd_pcm_t *pcm,
-			 snd_pcm_hw_params_t *hwparams,
-			 FILE *fp);
-
-bool
-gri_alsa_pick_acceptable_format (snd_pcm_t *pcm,
-				 snd_pcm_hw_params_t *hwparams,
-				 snd_pcm_format_t acceptable_formats[],
-				 unsigned nacceptable_formats,
-				 snd_pcm_format_t *selected_format,
-				 const char *error_msg_tag,
-				 bool verbose);
-
-
-#endif /* INCLUDED_GRI_ALSA_H */
diff --git a/gr-audio-alsa/src/qa_alsa.py b/gr-audio-alsa/src/qa_alsa.py
deleted file mode 100755
index 52dbfdc7a..000000000
--- a/gr-audio-alsa/src/qa_alsa.py
+++ /dev/null
@@ -1,40 +0,0 @@
-#!/usr/bin/env python
-#
-# Copyright 2005,2007,2010 Free Software Foundation, Inc.
-# 
-# This file is part of GNU Radio
-# 
-# GNU Radio is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 3, or (at your option)
-# any later version.
-# 
-# GNU Radio is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-# 
-# You should have received a copy of the GNU General Public License
-# along with GNU Radio; see the file COPYING.  If not, write to
-# the Free Software Foundation, Inc., 51 Franklin Street,
-# Boston, MA 02110-1301, USA.
-# 
-
-from gnuradio import gr, gr_unittest
-import audio_alsa
-
-class test_audio_alsa (gr_unittest.TestCase):
-
-    def setUp (self):
-        self.tb = gr.top_block ()
-
-    def tearDown (self):
-        self.tb = None
-
-    def test_000_nop (self):
-        """Just see if we can import the module...
-        They may not have ALSA drivers, etc.  Don't try to run anything"""
-        pass
-    
-if __name__ == '__main__':
-    gr_unittest.run(test_audio_alsa, "test_audio_alsa.xml")
diff --git a/gr-audio-alsa/src/run_tests.in b/gr-audio-alsa/src/run_tests.in
deleted file mode 100644
index fd3963068..000000000
--- a/gr-audio-alsa/src/run_tests.in
+++ /dev/null
@@ -1,10 +0,0 @@
-#!/bin/sh
-
-# 1st parameter is absolute path to component source directory
-# 2nd parameter is absolute path to component build directory
-# 3rd parameter is path to Python QA directory
-
-@top_builddir@/run_tests.sh \
-    @abs_top_srcdir@/gr-audio-alsa \
-    @abs_top_builddir@/gr-audio-alsa \
-    @srcdir@