Windows: Use SpeexDSP binary package

Remove bundled code and use vcpkg binary library instead.
author: João Valverde <j@v6e.pt> 2022-12-15 18:54:48 +0000
committer: João Valverde <j@v6e.pt> 2022-12-16 11:11:28 +0000
commit: ae148498649e38105607e9da2262fc276ffe2049 (patch)
tree: a603adaab2f0b334fb39eb72f0644a7ab91da838
parent: a4f56e90a66e81f85d8e5f81eb3e5781198311c3 (diff)
19 files changed, 44 insertions, 1977 deletions
diff --git a/CMakeLists.txt b/CMakeLists.txt
index 7abb22db86..8f4758ae64 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -1327,19 +1327,7 @@ find_package(DOXYGEN)
 
 # The SpeexDSP resampler is required iff building wireshark or sharkd.
 if(BUILD_wireshark OR BUILD_logray OR BUILD_sharkd)
-	# We don't provide a binary package for SpeexDSP in our repository.
-	# If using the repository don't bother searching for a system SpeexDSP
-	# installation and just use the bundled resampler code instead.
-	if (NOT USE_REPOSITORY)
-		find_package(SpeexDSP)
-	endif()
-	if(SpeexDSP_FOUND)
-		set(HAVE_SPEEXDSP 1)
-	else()
-		add_subdirectory(speexdsp)
-		set(SPEEXDSP_INCLUDE_DIRS "")
-		set(SPEEXDSP_LIBRARIES "speexresampler")
-	endif()
+	find_package(SpeexDSP REQUIRED)
 endif()
 
 # Generate the distribution tarball.
@@ -2171,6 +2159,9 @@ if(USE_REPOSITORY)
 			list (APPEND THIRD_PARTY_DLLS "${BROTLI_DLL_DIR}/${_dll}")
 		endforeach(_dll)
 	endif(BROTLI_FOUND)
+	if (SPEEXDSP_FOUND)
+		list (APPEND THIRD_PARTY_DLLS "${SPEEXDSP_DLL_DIR}/${SPEEXDSP_DLL}")
+	endif()
 
 	# With libs downloaded to c:/wireshark-win64-libs this currently
 	# (early 2018) expands to about 1900 characters.
diff --git a/cmake/modules/FindSpeexDSP.cmake b/cmake/modules/FindSpeexDSP.cmake
index 6c111111e5..cd6ada0c42 100644
--- a/cmake/modules/FindSpeexDSP.cmake
+++ b/cmake/modules/FindSpeexDSP.cmake
@@ -1,18 +1,26 @@
+# Find the system's SpeexDSP includes and library
 #
-# - Find speexdsp libraries
-#
-#  SPEEXDSP_INCLUDE_DIRS - where to find speexdsp headers.
-#  SPEEXDSP_LIBRARIES    - List of libraries when using speexdsp.
-#  SPEEXDSP_FOUND        - True if speexdsp is found.
+#  SPEEXDSP_INCLUDE_DIRS - where to find SpeexDSP headers
+#  SPEEXDSP_LIBRARIES    - List of libraries when using SpeexDSP
+#  SPEEXDSP_FOUND        - True if SpeexDSP found
+#  SPEEXDSP_DLL_DIR      - (Windows) Path to the SpeexDSP DLL
+#  SPEEXDSP_DLL          - (Windows) Name of the SpeexDSP DLL
+
+include(FindWSWinLibs)
+FindWSWinLibs("speexdsp-.*" "SPEEXDSP_HINTS")
+
+if(NOT WIN32)
+  find_package(PkgConfig)
+  pkg_search_module(PC_SPEEXDSP speexdsp)
+endif()
 
-find_package(PkgConfig QUIET)
-pkg_search_module(PC_SPEEXDSP QUIET speexdsp)
 
 find_path(SPEEXDSP_INCLUDE_DIR
   NAMES
     speex/speex_resampler.h
   HINTS
     ${PC_SPEEXDSP_INCLUDE_DIRS}
+    ${SPEEXDSP_HINTS}/include
 )
 
 find_library(SPEEXDSP_LIBRARY
@@ -20,16 +28,28 @@ find_library(SPEEXDSP_LIBRARY
     speexdsp
   HINTS
     ${PC_SPEEXDSP_LIBRARY_DIRS}
+    ${SPEEXDSP_HINTS}/lib
 )
 
 include(FindPackageHandleStandardArgs)
-find_package_handle_standard_args(SpeexDSP
-  REQUIRED_VARS   SPEEXDSP_LIBRARY SPEEXDSP_INCLUDE_DIR
-  VERSION_VAR     PC_SPEEXDSP_VERSION)
+find_package_handle_standard_args(SpeexDSP DEFAULT_MSG SPEEXDSP_LIBRARY SPEEXDSP_INCLUDE_DIR)
 
 if(SPEEXDSP_FOUND)
   set(SPEEXDSP_LIBRARIES ${SPEEXDSP_LIBRARY})
   set(SPEEXDSP_INCLUDE_DIRS ${SPEEXDSP_INCLUDE_DIR})
+  if(WIN32)
+    set(SPEEXDSP_DLL_DIR "${SPEEXDSP_HINTS}/bin"
+      CACHE PATH "Path to SpeexDSP DLL"
+    )
+    file(GLOB _speexdsp_dll RELATIVE "${SPEEXDSP_DLL_DIR}"
+      "${SPEEXDSP_DLL_DIR}/libspeexdsp.dll"
+    )
+    set(SPEEXDSP_DLL ${_speexdsp_dll}
+      # We're storing filenames only. Should we use STRING instead?
+      CACHE FILEPATH "SpeexDSP DLL file name"
+    )
+    mark_as_advanced(SPEEXDSP_DLL_DIR SPEEXDSP_DLL)
+  endif()
 else()
   set(SPEEXDSP_LIBRARIES)
   set(SPEEXDSP_INCLUDE_DIRS)
diff --git a/cmakeconfig.h.in b/cmakeconfig.h.in
index 9fbedea831..1b60bc23ad 100644
--- a/cmakeconfig.h.in
+++ b/cmakeconfig.h.in
@@ -235,9 +235,6 @@
 /* Define to 1 if you have the opus library. */
 #cmakedefine HAVE_OPUS 1
 
-/* Define to 1 if you have the speexdsp library. */
-#cmakedefine HAVE_SPEEXDSP 1
-
 /* Define to 1 if you have the lixbml2 library. */
 #cmakedefine HAVE_LIBXML2 1
 
diff --git a/docbook/release-notes.adoc b/docbook/release-notes.adoc
index 7730d5203a..7d3cf60f15 100644
--- a/docbook/release-notes.adoc
+++ b/docbook/release-notes.adoc
@@ -25,6 +25,9 @@ A new display filter feature for filtering raw bytes has been added.
 
 Display filter autocomplete is smarter about not suggesting invalid syntax.
 
+The Windows build has a new SpeexDSP external dependency (https://www.speex.org).
+The speex code that was previously bundled has been removed.
+
 Many other improvements have been made.
 See the “New and Updated Features” section below for more details.
 
diff --git a/doxygen.cfg.in b/doxygen.cfg.in
index 8b73ab5137..4175d09647 100644
--- a/doxygen.cfg.in
+++ b/doxygen.cfg.in
@@ -837,7 +837,6 @@ INPUT                  = @DOXYGEN_INPUT_DIRECTORY@/wireshark.dox \
                          @DOXYGEN_INPUT_DIRECTORY@/extcap \
                          @DOXYGEN_INPUT_DIRECTORY@/plugins \
                          @DOXYGEN_INPUT_DIRECTORY@/randpkt_core \
-                         @DOXYGEN_INPUT_DIRECTORY@/speexdsp \
                          @DOXYGEN_INPUT_DIRECTORY@/ui \
                          @DOXYGEN_INPUT_DIRECTORY@/wiretap \
                          @DOXYGEN_INPUT_DIRECTORY@/writecap \
diff --git a/packaging/nsis/CMakeLists.txt b/packaging/nsis/CMakeLists.txt
index 8d4b61513b..a6b2060ff6 100644
--- a/packaging/nsis/CMakeLists.txt
+++ b/packaging/nsis/CMakeLists.txt
@@ -177,6 +177,7 @@ if (BUILD_wireshark)
 			${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL}
 			${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL}
 			${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL}
+			${SPEEXDSP_DLL}
 			# Needed for mmdbresolve
 			${MAXMINDDB_DLL}
 		)
@@ -215,6 +216,7 @@ if (BUILD_logray)
 			${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL}
 			${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL}
 			${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL}
+			${SPEEXDSP_DLL}
 			# Needed for mmdbresolve
 			${MAXMINDDB_DLL}
 		)
diff --git a/packaging/wix/CMakeLists.txt b/packaging/wix/CMakeLists.txt
index a0925e9e71..6ce90ef2fd 100644
--- a/packaging/wix/CMakeLists.txt
+++ b/packaging/wix/CMakeLists.txt
@@ -138,6 +138,7 @@ foreach(_dll ${CARES_DLL} ${PCRE2_DLL} ${GCRYPT_DLLS}
 		${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL}
 		${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL}
 		${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL}
+		${SPEEXDSP_DLL}
 		# Required for mmdbresolve
 		${MAXMINDDB_DLL}
 	)
@@ -177,6 +178,7 @@ foreach(_dll ${CARES_DLL} ${PCRE2_DLL} ${GCRYPT_DLLS}
 		${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL}
 		${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL}
 		${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL}
+		${SPEEXDSP_DLL}
 		# mmdbresolve
 		${MAXMINDDB_DLL}
 	)
diff --git a/sharkd_session.c b/sharkd_session.c
index b34ab30f84..3f5414aee1 100644
--- a/sharkd_session.c
+++ b/sharkd_session.c
@@ -61,11 +61,7 @@
 #include <epan/addr_resolv.h>
 #include <epan/dissectors/packet-rtp.h>
 #include <ui/rtp_media.h>
-#ifdef HAVE_SPEEXDSP
-# include <speex/speex_resampler.h>
-#else
-# include "speexdsp/speex_resampler.h"
-#endif /* HAVE_SPEEXDSP */
+#include <speex/speex_resampler.h>
 
 #include <epan/maxmind_db.h>
 
diff --git a/speexdsp/CMakeLists.txt b/speexdsp/CMakeLists.txt
deleted file mode 100644
index 1667dfec28..0000000000
--- a/speexdsp/CMakeLists.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-# CMakeLists.txt
-#
-# Wireshark - Network traffic analyzer
-# By Gerald Combs <gerald@wireshark.org>
-# Copyright 1998 Gerald Combs
-#
-# SPDX-License-Identifier: GPL-2.0-or-later
-#
-
-add_library(speexresampler STATIC resample.c)
-
-set_target_properties(speexresampler PROPERTIES
-	LINK_FLAGS "${WS_LINK_FLAGS}"
-	FOLDER "Libs"
-)
diff --git a/speexdsp/README.txt b/speexdsp/README.txt
deleted file mode 100644
index ee8cd66e13..0000000000
--- a/speexdsp/README.txt
+++ /dev/null
@@ -1,2 +0,0 @@
-Copied from http://git.xiph.org/speexdsp.git c470e2e89a6ca75b507437467692cd684b71a526
-and modified to compile in the Wireshark source tree.
diff --git a/speexdsp/arch.h b/speexdsp/arch.h
deleted file mode 100644
index eaad6b9260..0000000000
--- a/speexdsp/arch.h
+++ /dev/null
@@ -1,237 +0,0 @@
-/* Copyright (C) 2003 Jean-Marc Valin */
-/**
-   @file arch.h
-   @brief Various architecture definitions Speex
-*/
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef ARCH_H
-#define ARCH_H
-
-/* A couple test to catch stupid option combinations */
-#ifdef FIXED_POINT
-
-#ifdef FLOATING_POINT
-#error You cannot compile as floating point and fixed point at the same time
-#endif
-#ifdef _USE_SSE
-#error SSE is only for floating-point
-#endif
-#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM))
-#error Make up your mind. What CPU do you have?
-#endif
-#ifdef VORBIS_PSYCHO
-#error Vorbis-psy model currently not implemented in fixed-point
-#endif
-
-#else
-
-#ifndef FLOATING_POINT
-#error You now need to define either FIXED_POINT or FLOATING_POINT
-#endif
-#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM)
-#error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions?
-#endif
-#ifdef FIXED_POINT_DEBUG
-#error "Don't you think enabling fixed-point is a good thing to do if you want to debug that?"
-#endif
-
-
-#endif
-
-#ifndef OUTSIDE_SPEEX
-#include "speex/speexdsp_types.h"
-#endif
-
-#ifndef ABS /* already defined by glib.h */
-#define ABS(x) ((x) < 0 ? (-(x)) : (x))      /**< Absolute integer value. */
-#endif
-#define ABS16(x) ((x) < 0 ? (-(x)) : (x))    /**< Absolute 16-bit value.  */
-#define MIN16(a,b) ((a) < (b) ? (a) : (b))   /**< Maximum 16-bit value.   */
-#define MAX16(a,b) ((a) > (b) ? (a) : (b))   /**< Maximum 16-bit value.   */
-#define ABS32(x) ((x) < 0 ? (-(x)) : (x))    /**< Absolute 32-bit value.  */
-#define MIN32(a,b) ((a) < (b) ? (a) : (b))   /**< Maximum 32-bit value.   */
-#define MAX32(a,b) ((a) > (b) ? (a) : (b))   /**< Maximum 32-bit value.   */
-
-#ifdef FIXED_POINT
-
-typedef spx_int16_t spx_word16_t;
-typedef spx_int32_t spx_word32_t;
-typedef spx_word32_t spx_mem_t;
-typedef spx_word16_t spx_coef_t;
-typedef spx_word16_t spx_lsp_t;
-typedef spx_word32_t spx_sig_t;
-
-#define Q15ONE 32767
-
-#define LPC_SCALING  8192
-#define SIG_SCALING  16384
-#define LSP_SCALING  8192.
-#define GAMMA_SCALING 32768.
-#define GAIN_SCALING 64
-#define GAIN_SCALING_1 0.015625
-
-#define LPC_SHIFT    13
-#define LSP_SHIFT    13
-#define SIG_SHIFT    14
-#define GAIN_SHIFT   6
-
-#define WORD2INT(x) ((spx_int16_t)((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x))))
-
-#define VERY_SMALL 0
-#define VERY_LARGE32 ((spx_word32_t)2147483647)
-#define VERY_LARGE16 ((spx_word16_t)32767)
-#define Q15_ONE ((spx_word16_t)32767)
-
-
-#ifdef FIXED_DEBUG
-#include "fixed_debug.h"
-#else
-
-#include "fixed_generic.h"
-
-#ifdef ARM5E_ASM
-#include "fixed_arm5e.h"
-#elif defined (ARM4_ASM)
-#include "fixed_arm4.h"
-#elif defined (BFIN_ASM)
-#include "fixed_bfin.h"
-#endif
-
-#endif
-
-
-#else
-
-typedef float spx_mem_t;
-typedef float spx_coef_t;
-typedef float spx_lsp_t;
-typedef float spx_sig_t;
-typedef float spx_word16_t;
-typedef float spx_word32_t;
-
-#define Q15ONE 1.0f
-#define LPC_SCALING  1.f
-#define SIG_SCALING  1.f
-#define LSP_SCALING  1.f
-#define GAMMA_SCALING 1.f
-#define GAIN_SCALING 1.f
-#define GAIN_SCALING_1 1.f
-
-
-#define VERY_SMALL 1e-15f
-#define VERY_LARGE32 1e15f
-#define VERY_LARGE16 1e15f
-#define Q15_ONE ((spx_word16_t)1.f)
-
-#define QCONST16(x,bits) (x)
-#define QCONST32(x,bits) (x)
-
-#define NEG16(x) (-(x))
-#define NEG32(x) (-(x))
-#define EXTRACT16(x) (x)
-#define EXTEND32(x) (x)
-#define SHR16(a,shift) (a)
-#define SHL16(a,shift) (a)
-#define SHR32(a,shift) (a)
-#define SHL32(a,shift) (a)
-#define PSHR16(a,shift) (a)
-#define PSHR32(a,shift) ((spx_word16_t)(a))
-#define VSHR32(a,shift) (a)
-#define SATURATE16(x,a) (x)
-#define SATURATE32(x,a) (x)
-#define SATURATE32PSHR(x,shift,a) (x)
-
-#define PSHR(a,shift)       (a)
-#define SHR(a,shift)       (a)
-#define SHL(a,shift)       (a)
-#define SATURATE(x,a) (x)
-
-#define ADD16(a,b) ((a)+(b))
-#define SUB16(a,b) ((a)-(b))
-#define ADD32(a,b) ((a)+(b))
-#define SUB32(a,b) ((a)-(b))
-#define MULT16_16_16(a,b)     ((a)*(b))
-#define MULT16_16(a,b)     ((spx_word32_t)(a)*(spx_word32_t)(b))
-#define MAC16_16(c,a,b)     ((c)+(spx_word32_t)(a)*(spx_word32_t)(b))
-
-#define MULT16_32_Q11(a,b)     ((a)*(b))
-#define MULT16_32_Q13(a,b)     ((a)*(b))
-#define MULT16_32_Q14(a,b)     ((a)*(b))
-#define MULT16_32_Q15(a,b)     ((spx_word32_t)((a)*(b)))
-#define MULT16_32_P15(a,b)     ((a)*(b))
-
-#define MAC16_32_Q11(c,a,b)     ((c)+(a)*(b))
-#define MAC16_32_Q15(c,a,b)     ((c)+(a)*(b))
-
-#define MAC16_16_Q11(c,a,b)     ((c)+(a)*(b))
-#define MAC16_16_Q13(c,a,b)     ((c)+(a)*(b))
-#define MAC16_16_P13(c,a,b)     ((c)+(a)*(b))
-#define MULT16_16_Q11_32(a,b)     ((a)*(b))
-#define MULT16_16_Q13(a,b)     ((a)*(b))
-#define MULT16_16_Q14(a,b)     ((a)*(b))
-#define MULT16_16_Q15(a,b)     ((a)*(b))
-#define MULT16_16_P15(a,b)     ((a)*(b))
-#define MULT16_16_P13(a,b)     ((a)*(b))
-#define MULT16_16_P14(a,b)     ((a)*(b))
-
-#define DIV32_16(a,b)     (((spx_word32_t)(a))/(spx_word16_t)(b))
-#define PDIV32_16(a,b)     (((spx_word32_t)(a))/(spx_word16_t)(b))
-#define DIV32(a,b)     (((spx_word32_t)(a))/(spx_word32_t)(b))
-#define PDIV32(a,b)     (((spx_word32_t)(a))/(spx_word32_t)(b))
-
-#define WORD2INT(x) ((spx_int16_t)((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x)))))
-
-#endif
-
-
-#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
-
-/* 2 on TI C5x DSP */
-#define BYTES_PER_CHAR 2
-#define BITS_PER_CHAR 16
-#define LOG2_BITS_PER_CHAR 4
-
-#else
-
-#define BYTES_PER_CHAR 1
-#define BITS_PER_CHAR 8
-#define LOG2_BITS_PER_CHAR 3
-
-#endif
-
-
-
-#ifdef FIXED_DEBUG
-extern long long spx_mips;
-#endif
-
-
-#endif
diff --git a/speexdsp/resample.c b/speexdsp/resample.c
deleted file mode 100644
index 32197fe5f6..0000000000
--- a/speexdsp/resample.c
+++ /dev/null
@@ -1,1211 +0,0 @@
-/* Copyright (C) 2007-2008 Jean-Marc Valin
-   Copyright (C) 2008      Thorvald Natvig
-
-   File: resample.c
-   Arbitrary resampling code
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions are
-   met:
-
-   1. Redistributions of source code must retain the above copyright notice,
-   this list of conditions and the following disclaimer.
-
-   2. Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   3. The name of the author may not be used to endorse or promote products
-   derived from this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
-   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
-   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
-   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
-   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
-   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
-   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
-   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-   POSSIBILITY OF SUCH DAMAGE.
-*/
-
-/*
-   The design goals of this code are:
-      - Very fast algorithm
-      - SIMD-friendly algorithm
-      - Low memory requirement
-      - Good *perceptual* quality (and not best SNR)
-
-   Warning: This resampler is relatively new. Although I think I got rid of
-   all the major bugs and I don't expect the API to change anymore, there
-   may be something I've missed. So use with caution.
-
-   This algorithm is based on this original resampling algorithm:
-   Smith, Julius O. Digital Audio Resampling Home Page
-   Center for Computer Research in Music and Acoustics (CCRMA),
-   Stanford University, 2007.
-   Web published at http://www-ccrma.stanford.edu/~jos/resample/.
-
-   There is one main difference, though. This resampler uses cubic
-   interpolation instead of linear interpolation in the above paper. This
-   makes the table much smaller and makes it possible to compute that table
-   on a per-stream basis. In turn, being able to tweak the table for each
-   stream makes it possible to both reduce complexity on simple ratios
-   (e.g. 2/3), and get rid of the rounding operations in the inner loop.
-   The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
-*/
-
-#include <config.h>
-
-#define OUTSIDE_SPEEX 1
-#define FLOATING_POINT 1
-
-#ifdef OUTSIDE_SPEEX
-#include <glib.h>
-static void *speex_alloc (size_t size) {return g_malloc0(size);}
-static void *speex_realloc (void *ptr, size_t size) {return g_realloc(ptr, size);}
-static void speex_free (void *ptr) {g_free(ptr);}
-#include "speex_resampler.h"
-#include "arch.h"
-#else /* OUTSIDE_SPEEX */
-
-#include "speex/speex_resampler.h"
-#include "arch.h"
-#include "os_support.h"
-#endif /* OUTSIDE_SPEEX */
-
-#include "stack_alloc.h"
-#include <math.h>
-#include <limits.h>
-
-#include "ws_attributes.h"
-
-#ifndef M_PI
-#define M_PI 3.14159265358979323846
-#endif
-
-#define IMAX(a,b) ((a) > (b) ? (a) : (b))
-#define IMIN(a,b) ((a) < (b) ? (a) : (b))
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-#ifdef _USE_SSE
-#include "resample_sse.h"
-#endif
-
-#ifdef _USE_NEON
-#include "resample_neon.h"
-#endif
-
-/* Numer of elements to allocate on the stack */
-#ifdef VAR_ARRAYS
-#define FIXED_STACK_ALLOC 8192
-#else
-#define FIXED_STACK_ALLOC 1024
-#endif
-
-typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
-
-struct SpeexResamplerState_ {
-   spx_uint32_t in_rate;
-   spx_uint32_t out_rate;
-   spx_uint32_t num_rate;
-   spx_uint32_t den_rate;
-
-   int    quality;
-   spx_uint32_t nb_channels;
-   spx_uint32_t filt_len;
-   spx_uint32_t mem_alloc_size;
-   spx_uint32_t buffer_size;
-   int          int_advance;
-   int          frac_advance;
-   float  cutoff;
-   spx_uint32_t oversample;
-   int          initialised;
-   int          started;
-
-   /* These are per-channel */
-   spx_int32_t  *last_sample;
-   spx_uint32_t *samp_frac_num;
-   spx_uint32_t *magic_samples;
-
-   spx_word16_t *mem;
-   spx_word16_t *sinc_table;
-   spx_uint32_t sinc_table_length;
-   resampler_basic_func resampler_ptr;
-
-   int    in_stride;
-   int    out_stride;
-} ;
-
-static const double kaiser12_table[68] = {
-   0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
-   0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
-   0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
-   0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
-   0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
-   0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
-   0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
-   0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
-   0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
-   0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
-   0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
-   0.00001000, 0.00000000};
-/*
-static const double kaiser12_table[36] = {
-   0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
-   0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
-   0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
-   0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
-   0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
-   0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
-*/
-static const double kaiser10_table[36] = {
-   0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
-   0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
-   0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
-   0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
-   0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
-   0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};
-
-static const double kaiser8_table[36] = {
-   0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
-   0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
-   0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
-   0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
-   0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
-   0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
-
-static const double kaiser6_table[36] = {
-   0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
-   0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
-   0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
-   0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
-   0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
-   0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};
-
-struct FuncDef {
-   const double *table;
-   int oversample;
-};
-
-static const struct FuncDef _KAISER12 = {kaiser12_table, 64};
-#define KAISER12 (&_KAISER12)
-/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
-#define KAISER12 (&_KAISER12)*/
-static const struct FuncDef _KAISER10 = {kaiser10_table, 32};
-#define KAISER10 (&_KAISER10)
-static const struct FuncDef _KAISER8 = {kaiser8_table, 32};
-#define KAISER8 (&_KAISER8)
-static const struct FuncDef _KAISER6 = {kaiser6_table, 32};
-#define KAISER6 (&_KAISER6)
-
-struct QualityMapping {
-   int base_length;
-   int oversample;
-   float downsample_bandwidth;
-   float upsample_bandwidth;
-   const struct FuncDef *window_func;
-};
-
-
-/* This table maps conversion quality to internal parameters. There are two
-   reasons that explain why the up-sampling bandwidth is larger than the
-   down-sampling bandwidth:
-   1) When up-sampling, we can assume that the spectrum is already attenuated
-      close to the Nyquist rate (from an A/D or a previous resampling filter)
-   2) Any aliasing that occurs very close to the Nyquist rate will be masked
-      by the sinusoids/noise just below the Nyquist rate (guaranteed only for
-      up-sampling).
-*/
-static const struct QualityMapping quality_map[11] = {
-   {  8,  4, 0.830f, 0.860f, KAISER6 }, /* Q0 */
-   { 16,  4, 0.850f, 0.880f, KAISER6 }, /* Q1 */
-   { 32,  4, 0.882f, 0.910f, KAISER6 }, /* Q2 */  /* 82.3% cutoff ( ~60 dB stop) 6  */
-   { 48,  8, 0.895f, 0.917f, KAISER8 }, /* Q3 */  /* 84.9% cutoff ( ~80 dB stop) 8  */
-   { 64,  8, 0.921f, 0.940f, KAISER8 }, /* Q4 */  /* 88.7% cutoff ( ~80 dB stop) 8  */
-   { 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */  /* 89.1% cutoff (~100 dB stop) 10 */
-   { 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */  /* 91.5% cutoff (~100 dB stop) 10 */
-   {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */  /* 93.1% cutoff (~100 dB stop) 10 */
-   {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */  /* 94.5% cutoff (~100 dB stop) 10 */
-   {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */  /* 95.5% cutoff (~100 dB stop) 10 */
-   {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
-};
-/*8,24,40,56,80,104,128,160,200,256,320*/
-static double compute_func(float x, const struct FuncDef *func)
-{
-   float y, frac;
-   double interp[4];
-   int ind;
-   y = x*func->oversample;
-   ind = (int)floor(y);
-   frac = (y-ind);
-   /* CSE with handle the repeated powers */
-   interp[3] =  -0.1666666667*frac + 0.1666666667*(frac*frac*frac);
-   interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac);
-   /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
-   interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac);
-   /* Just to make sure we don't have rounding problems */
-   interp[1] = 1.f-interp[3]-interp[2]-interp[0];
-
-   /*sum = frac*accum[1] + (1-frac)*accum[2];*/
-   return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];
-}
-
-#if 0
-#include <stdio.h>
-int main(int argc, char **argv)
-{
-   int i;
-   for (i=0;i<256;i++)
-   {
-      printf ("%f\n", compute_func(i/256., KAISER12));
-   }
-   return 0;
-}
-#endif
-
-DIAG_OFF_CLANG(self-assign) /* SATURATE32PSHR */
-
-#ifdef FIXED_POINT
-/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
-{
-   /*fprintf (stderr, "%f ", x);*/
-   float xx = x * cutoff;
-   if (fabs(x)<1e-6f)
-      return WORD2INT(32768.*cutoff);
-   else if (fabs(x) > .5f*N)
-      return 0;
-   /*FIXME: Can it really be any slower than this? */
-   return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));
-}
-#else
-/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
-{
-   /*fprintf (stderr, "%f ", x);*/
-   float xx = x * cutoff;
-   if (fabs(x)<1e-6)
-      return cutoff;
-   else if (fabs(x) > .5*N)
-      return 0;
-   /*FIXME: Can it really be any slower than this? */
-   return (spx_word16_t)(cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func((float)fabs(2.*x/N), window_func));
-}
-#endif
-
-#ifdef FIXED_POINT
-static void cubic_coef(spx_word16_t x, spx_word16_t interp[4])
-{
-   /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
-   but I know it's MMSE-optimal on a sinc */
-   spx_word16_t x2, x3;
-   x2 = MULT16_16_P15(x, x);
-   x3 = MULT16_16_P15(x, x2);
-   interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15);
-   interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1));
-   interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15);
-   /* Just to make sure we don't have rounding problems */
-   interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3];
-   if (interp[2]<32767)
-      interp[2]+=1;
-}
-#else
-static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4])
-{
-   /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
-   but I know it's MMSE-optimal on a sinc */
-   interp[0] =  -0.16667f*frac + 0.16667f*frac*frac*frac;
-   interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac;
-   /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
-   interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac;
-   /* Just to make sure we don't have rounding problems */
-   interp[2] = (spx_word16_t)(1.-interp[0]-interp[1]-interp[3]);
-}
-#endif
-
-static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
-   const int N = st->filt_len;
-   int out_sample = 0;
-   int last_sample = st->last_sample[channel_index];
-   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-   const spx_word16_t *sinc_table = st->sinc_table;
-   const int out_stride = st->out_stride;
-   const int int_advance = st->int_advance;
-   const int frac_advance = st->frac_advance;
-   const spx_uint32_t den_rate = st->den_rate;
-   spx_word32_t sum;
-
-   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
-   {
-      const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
-      const spx_word16_t *iptr = & in[last_sample];
-
-#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
-      int j;
-      sum = 0;
-      for(j=0;j<N;j++) sum += MULT16_16(sinct[j], iptr[j]);
-
-/*    This code is slower on most DSPs which have only 2 accumulators.
-      Plus this forces truncation to 32 bits and you lose the HW guard bits.
-      I think we can trust the compiler and let it vectorize and/or unroll itself.
-      spx_word32_t accum[4] = {0,0,0,0};
-      for(j=0;j<N;j+=4) {
-        accum[0] += MULT16_16(sinct[j], iptr[j]);
-        accum[1] += MULT16_16(sinct[j+1], iptr[j+1]);
-        accum[2] += MULT16_16(sinct[j+2], iptr[j+2]);
-        accum[3] += MULT16_16(sinct[j+3], iptr[j+3]);
-      }
-      sum = accum[0] + accum[1] + accum[2] + accum[3];
-*/
-      sum = SATURATE32PSHR(sum, 15, 32767);
-#else
-      sum = inner_product_single(sinct, iptr, N);
-#endif
-
-      out[out_stride * out_sample++] = sum;
-      last_sample += int_advance;
-      samp_frac_num += frac_advance;
-      if (samp_frac_num >= den_rate)
-      {
-         samp_frac_num -= den_rate;
-         last_sample++;
-      }
-   }
-
-   st->last_sample[channel_index] = last_sample;
-   st->samp_frac_num[channel_index] = samp_frac_num;
-   return out_sample;
-}
-
-#ifdef FIXED_POINT
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
-   const int N = st->filt_len;
-   int out_sample = 0;
-   int last_sample = st->last_sample[channel_index];
-   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-   const spx_word16_t *sinc_table = st->sinc_table;
-   const int out_stride = st->out_stride;
-   const int int_advance = st->int_advance;
-   const int frac_advance = st->frac_advance;
-   const spx_uint32_t den_rate = st->den_rate;
-   double sum;
-
-   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
-   {
-      const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
-      const spx_word16_t *iptr = & in[last_sample];
-
-#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
-      int j;
-      double accum[4] = {0,0,0,0};
-
-      for(j=0;j<N;j+=4) {
-        accum[0] += sinct[j]*iptr[j];
-        accum[1] += sinct[j+1]*iptr[j+1];
-        accum[2] += sinct[j+2]*iptr[j+2];
-        accum[3] += sinct[j+3]*iptr[j+3];
-      }
-      sum = accum[0] + accum[1] + accum[2] + accum[3];
-#else
-      sum = inner_product_double(sinct, iptr, N);
-#endif
-
-      out[out_stride * out_sample++] = PSHR32(sum, 15);
-      last_sample += int_advance;
-      samp_frac_num += frac_advance;
-      if (samp_frac_num >= den_rate)
-      {
-         samp_frac_num -= den_rate;
-         last_sample++;
-      }
-   }
-
-   st->last_sample[channel_index] = last_sample;
-   st->samp_frac_num[channel_index] = samp_frac_num;
-   return out_sample;
-}
-#endif
-
-static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
-   const int N = st->filt_len;
-   int out_sample = 0;
-   int last_sample = st->last_sample[channel_index];
-   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-   const int out_stride = st->out_stride;
-   const int int_advance = st->int_advance;
-   const int frac_advance = st->frac_advance;
-   const spx_uint32_t den_rate = st->den_rate;
-   spx_word32_t sum;
-
-   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
-   {
-      const spx_word16_t *iptr = & in[last_sample];
-
-      const int offset = samp_frac_num*st->oversample/st->den_rate;
-#ifdef FIXED_POINT
-      const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
-#else
-      const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
-#endif
-      spx_word16_t interp[4];
-
-
-#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
-      int j;
-      spx_word32_t accum[4] = {0,0,0,0};
-
-      for(j=0;j<N;j++) {
-        const spx_word16_t curr_in=iptr[j];
-        accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
-        accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
-        accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
-        accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
-      }
-
-      cubic_coef(frac, interp);
-      sum = MULT16_32_Q15(interp[0],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1));
-      sum = SATURATE32PSHR(sum, 15, 32767);
-#else
-      cubic_coef(frac, interp);
-      sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
-#endif
-
-      out[out_stride * out_sample++] = sum;
-      last_sample += int_advance;
-      samp_frac_num += frac_advance;
-      if (samp_frac_num >= den_rate)
-      {
-         samp_frac_num -= den_rate;
-         last_sample++;
-      }
-   }
-
-   st->last_sample[channel_index] = last_sample;
-   st->samp_frac_num[channel_index] = samp_frac_num;
-   return out_sample;
-}
-
-#ifdef FIXED_POINT
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
-   const int N = st->filt_len;
-   int out_sample = 0;
-   int last_sample = st->last_sample[channel_index];
-   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-   const int out_stride = st->out_stride;
-   const int int_advance = st->int_advance;
-   const int frac_advance = st->frac_advance;
-   const spx_uint32_t den_rate = st->den_rate;
-   spx_word32_t sum;
-
-   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
-   {
-      const spx_word16_t *iptr = & in[last_sample];
-
-      const int offset = samp_frac_num*st->oversample/st->den_rate;
-#ifdef FIXED_POINT
-      const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
-#else
-      const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
-#endif
-      spx_word16_t interp[4];
-
-
-#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
-      int j;
-      double accum[4] = {0,0,0,0};
-
-      for(j=0;j<N;j++) {
-        const double curr_in=iptr[j];
-        accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
-        accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
-        accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
-        accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
-      }
-
-      cubic_coef(frac, interp);
-      sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
-#else
-      cubic_coef(frac, interp);
-      sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
-#endif
-
-      out[out_stride * out_sample++] = PSHR32(sum,15);
-      last_sample += int_advance;
-      samp_frac_num += frac_advance;
-      if (samp_frac_num >= den_rate)
-      {
-         samp_frac_num -= den_rate;
-         last_sample++;
-      }
-   }
-
-   st->last_sample[channel_index] = last_sample;
-   st->samp_frac_num[channel_index] = samp_frac_num;
-   return out_sample;
-}
-#endif
-
-/* This resampler is used to produce zero output in situations where memory
-   for the filter could not be allocated.  The expected numbers of input and
-   output samples are still processed so that callers failing to check error
-   codes are not surprised, possibly getting into infinite loops. */
-static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in _U_, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
-   int out_sample = 0;
-   int last_sample = st->last_sample[channel_index];
-   spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-   const int out_stride = st->out_stride;
-   const int int_advance = st->int_advance;
-   const int frac_advance = st->frac_advance;
-   const spx_uint32_t den_rate = st->den_rate;
-
-   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
-   {
-      out[out_stride * out_sample++] = 0;
-      last_sample += int_advance;
-      samp_frac_num += frac_advance;
-      if (samp_frac_num >= den_rate)
-      {
-         samp_frac_num -= den_rate;
-         last_sample++;
-      }
-   }
-
-   st->last_sample[channel_index] = last_sample;
-   st->samp_frac_num[channel_index] = samp_frac_num;
-   return out_sample;
-}
-
-static int update_filter(SpeexResamplerState *st)
-{
-   spx_uint32_t old_length = st->filt_len;
-   spx_uint32_t old_alloc_size = st->mem_alloc_size;
-   int use_direct;
-   spx_uint32_t min_sinc_table_length;
-   spx_uint32_t min_alloc_size;
-
-   st->int_advance = st->num_rate/st->den_rate;
-   st->frac_advance = st->num_rate%st->den_rate;
-   st->oversample = quality_map[st->quality].oversample;
-   st->filt_len = quality_map[st->quality].base_length;
-
-   if (st->num_rate > st->den_rate)
-   {
-      /* down-sampling */
-      st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
-      /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
-      st->filt_len = st->filt_len*st->num_rate / st->den_rate;
-      /* Round up to make sure we have a multiple of 8 for SSE */
-      st->filt_len = ((st->filt_len-1)&(~0x7))+8;
-      if (2*st->den_rate < st->num_rate)
-         st->oversample >>= 1;
-      if (4*st->den_rate < st->num_rate)
-         st->oversample >>= 1;
-      if (8*st->den_rate < st->num_rate)
-         st->oversample >>= 1;
-      if (16*st->den_rate < st->num_rate)
-         st->oversample >>= 1;
-      if (st->oversample < 1)
-         st->oversample = 1;
-   } else {
-      /* up-sampling */
-      st->cutoff = quality_map[st->quality].upsample_bandwidth;
-   }
-
-   /* Choose the resampling type that requires the least amount of memory */
-#ifdef RESAMPLE_FULL_SINC_TABLE
-   use_direct = 1;
-   if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len)
-      goto fail;
-#else
-   use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8
-                && INT_MAX/(spx_uint32_t)sizeof(spx_word16_t)/st->den_rate >= st->filt_len;
-#endif
-   if (use_direct)
-   {
-      min_sinc_table_length = st->filt_len*st->den_rate;
-   } else {
-      if ((INT_MAX/(spx_uint32_t)sizeof(spx_word16_t)-8)/st->oversample < st->filt_len)
-         goto fail;
-
-      min_sinc_table_length = st->filt_len*st->oversample+8;
-   }
-   if (st->sinc_table_length < min_sinc_table_length)
-   {
-      spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,min_sinc_table_length*sizeof(spx_word16_t));
-      if (!sinc_table)
-         goto fail;
-
-      st->sinc_table = sinc_table;
-      st->sinc_table_length = min_sinc_table_length;
-   }
-   if (use_direct)
-   {
-      spx_uint32_t i;
-      for (i=0;i<st->den_rate;i++)
-      {
-         spx_uint32_t j;
-         for (j=0;j<st->filt_len;j++)
-         {
-            st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
-         }
-      }
-#ifdef FIXED_POINT
-      st->resampler_ptr = resampler_basic_direct_single;
-#else
-      if (st->quality>8)
-         st->resampler_ptr = resampler_basic_direct_double;
-      else
-         st->resampler_ptr = resampler_basic_direct_single;
-#endif
-      /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
-   } else {
-      spx_int32_t i;
-      for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
-         st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
-#ifdef FIXED_POINT
-      st->resampler_ptr = resampler_basic_interpolate_single;
-#else
-      if (st->quality>8)
-         st->resampler_ptr = resampler_basic_interpolate_double;
-      else
-         st->resampler_ptr = resampler_basic_interpolate_single;
-#endif
-      /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
-   }
-
-   /* Here's the place where we update the filter memory to take into account
-      the change in filter length. It's probably the messiest part of the code
-      due to handling of lots of corner cases. */
-
-   /* Adding buffer_size to filt_len won't overflow here because filt_len
-      could be multiplied by sizeof(spx_word16_t) above. */
-   min_alloc_size = st->filt_len-1 + st->buffer_size;
-   if (min_alloc_size > st->mem_alloc_size)
-   {
-      spx_word16_t *mem;
-      if (INT_MAX/(spx_uint32_t)sizeof(spx_word16_t)/st->nb_channels < min_alloc_size)
-          goto fail;
-      else if (!(mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*min_alloc_size * sizeof(*mem))))
-          goto fail;
-
-      st->mem = mem;
-      st->mem_alloc_size = min_alloc_size;
-   }
-   if (!st->started)
-   {
-      spx_uint32_t i;
-      for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
-         st->mem[i] = 0;
-      /*speex_warning("reinit filter");*/
-   } else if (st->filt_len > old_length)
-   {
-      spx_uint32_t i;
-      /* Increase the filter length */
-      /*speex_warning("increase filter size");*/
-      for (i=st->nb_channels;i--;)
-      {
-         spx_uint32_t j;
-         spx_uint32_t olen = old_length;
-         /*if (st->magic_samples[i])*/
-         {
-            /* Try and remove the magic samples as if nothing had happened */
-
-            /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
-            olen = old_length + 2*st->magic_samples[i];
-            for (j=old_length-1+st->magic_samples[i];j--;)
-               st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
-            for (j=0;j<st->magic_samples[i];j++)
-               st->mem[i*st->mem_alloc_size+j] = 0;
-            st->magic_samples[i] = 0;
-         }
-         if (st->filt_len > olen)
-         {
-            /* If the new filter length is still bigger than the "augmented" length */
-            /* Copy data going backward */
-            for (j=0;j<olen-1;j++)
-               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
-            /* Then put zeros for lack of anything better */
-            for (;j<st->filt_len-1;j++)
-               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
-            /* Adjust last_sample */
-            st->last_sample[i] += (st->filt_len - olen)/2;
-         } else {
-            /* Put back some of the magic! */
-            st->magic_samples[i] = (olen - st->filt_len)/2;
-            for (j=0;j<st->filt_len-1+st->magic_samples[i];j++)
-               st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
-         }
-      }
-   } else if (st->filt_len < old_length)
-   {
-      spx_uint32_t i;
-      /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
-         samples so they can be used directly as input the next time(s) */
-      for (i=0;i<st->nb_channels;i++)
-      {
-         spx_uint32_t j;
-         spx_uint32_t old_magic = st->magic_samples[i];
-         st->magic_samples[i] = (old_length - st->filt_len)/2;
-         /* We must copy some of the memory that's no longer used */
-         /* Copy data going backward */
-         for (j=0;j<st->filt_len-1+st->magic_samples[i]+old_magic;j++)
-            st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
-         st->magic_samples[i] += old_magic;
-      }
-   }
-   return RESAMPLER_ERR_SUCCESS;
-
-fail:
-   st->resampler_ptr = resampler_basic_zero;
-   /* st->mem may still contain consumed input samples for the filter.
-      Restore filt_len so that filt_len - 1 still points to the position after
-      the last of these samples. */
-   st->filt_len = old_length;
-   return RESAMPLER_ERR_ALLOC_FAILED;
-}
-
-EXPORT SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
-{
-   return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err);
-}
-
-EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
-{
-   spx_uint32_t i;
-   SpeexResamplerState *st;
-   int filter_err;
-
-   if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || quality > 10 || quality < 0)
-   {
-      if (err)
-         *err = RESAMPLER_ERR_INVALID_ARG;
-      return NULL;
-   }
-   st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
-   st->initialised = 0;
-   st->started = 0;
-   st->in_rate = 0;
-   st->out_rate = 0;
-   st->num_rate = 0;
-   st->den_rate = 0;
-   st->quality = -1;
-   st->sinc_table_length = 0;
-   st->mem_alloc_size = 0;
-   st->filt_len = 0;
-   st->mem = 0;
-   st->resampler_ptr = 0;
-
-   st->cutoff = 1.f;
-   st->nb_channels = nb_channels;
-   st->in_stride = 1;
-   st->out_stride = 1;
-
-   st->buffer_size = 160;
-
-   /* Per channel data */
-   st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t));
-   st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
-   st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
-   for (i=0;i<nb_channels;i++)
-   {
-      st->last_sample[i] = 0;
-      st->magic_samples[i] = 0;
-      st->samp_frac_num[i] = 0;
-   }
-
-   speex_resampler_set_quality(st, quality);
-   speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
-
-   filter_err = update_filter(st);
-   if (filter_err == RESAMPLER_ERR_SUCCESS)
-   {
-      st->initialised = 1;
-   } else {
-      speex_resampler_destroy(st);
-      st = NULL;
-   }
-   if (err)
-      *err = filter_err;
-
-   return st;
-}
-
-EXPORT void speex_resampler_destroy(SpeexResamplerState *st)
-{
-   speex_free(st->mem);
-   speex_free(st->sinc_table);
-   speex_free(st->last_sample);
-   speex_free(st->magic_samples);
-   speex_free(st->samp_frac_num);
-   speex_free(st);
-}
-
-static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t channel_index, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
-   int j=0;
-   const int N = st->filt_len;
-   int out_sample = 0;
-   spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
-   spx_uint32_t ilen;
-
-   st->started = 1;
-
-   /* Call the right resampler through the function ptr */
-   out_sample = st->resampler_ptr(st, channel_index, mem, in_len, out, out_len);
-
-   if (st->last_sample[channel_index] < (spx_int32_t)*in_len)
-      *in_len = st->last_sample[channel_index];
-   *out_len = out_sample;
-   st->last_sample[channel_index] -= *in_len;
-
-   ilen = *in_len;
-
-   for(j=0;j<N-1;++j)
-     mem[j] = mem[j+ilen];
-
-   return RESAMPLER_ERR_SUCCESS;
-}
-
-static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_index, spx_word16_t **out, spx_uint32_t out_len) {
-   spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
-   spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
-   const int N = st->filt_len;
-
-   speex_resampler_process_native(st, channel_index, &tmp_in_len, *out, &out_len);
-
-   st->magic_samples[channel_index] -= tmp_in_len;
-
-   /* If we couldn't process all "magic" input samples, save the rest for next time */
-   if (st->magic_samples[channel_index])
-   {
-      spx_uint32_t i;
-      for (i=0;i<st->magic_samples[channel_index];i++)
-         mem[N-1+i]=mem[N-1+i+tmp_in_len];
-   }
-   *out += out_len*st->out_stride;
-   return out_len;
-}
-
-#ifdef FIXED_POINT
-EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
-#else
-EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
-#endif
-{
-   spx_uint32_t j;
-   spx_uint32_t ilen = *in_len;
-   spx_uint32_t olen = *out_len;
-   spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
-   const int filt_offs = st->filt_len - 1;
-   const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
-   const int istride = st->in_stride;
-
-   if (st->magic_samples[channel_index])
-      olen -= speex_resampler_magic(st, channel_index, &out, olen);
-   if (! st->magic_samples[channel_index]) {
-      while (ilen && olen) {
-        spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
-        spx_uint32_t ochunk = olen;
-
-        if (in) {
-           for(j=0;j<ichunk;++j)
-              x[j+filt_offs]=in[j*istride];
-        } else {
-          for(j=0;j<ichunk;++j)
-            x[j+filt_offs]=0;
-        }
-        speex_resampler_process_native(st, channel_index, &ichunk, out, &ochunk);
-        ilen -= ichunk;
-        olen -= ochunk;
-        out += ochunk * st->out_stride;
-        if (in)
-           in += ichunk * istride;
-      }
-   }
-   *in_len -= ilen;
-   *out_len -= olen;
-   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
-}
-
-#ifdef FIXED_POINT
-EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
-#else
-EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
-#endif
-{
-   spx_uint32_t j;
-   const int istride_save = st->in_stride;
-   const int ostride_save = st->out_stride;
-   spx_uint32_t ilen = *in_len;
-   spx_uint32_t olen = *out_len;
-   spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
-   const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
-#ifdef VAR_ARRAYS
-   const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
-   VARDECL(spx_word16_t *ystack);
-   ALLOC(ystack, ylen, spx_word16_t);
-#else
-   const unsigned int ylen = FIXED_STACK_ALLOC;
-   spx_word16_t ystack[FIXED_STACK_ALLOC];
-#endif
-
-   st->out_stride = 1;
-
-   while (ilen && olen) {
-     spx_word16_t *y = ystack;
-     spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
-     spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
-     spx_uint32_t omagic = 0;
-
-     if (st->magic_samples[channel_index]) {
-       omagic = speex_resampler_magic(st, channel_index, &y, ochunk);
-       ochunk -= omagic;
-       olen -= omagic;
-     }
-     if (! st->magic_samples[channel_index]) {
-       if (in) {
-         for(j=0;j<ichunk;++j)
-#ifdef FIXED_POINT
-           x[j+st->filt_len-1]=WORD2INT(in[j*istride_save]);
-#else
-           x[j+st->filt_len-1]=in[j*istride_save];
-#endif
-       } else {
-         for(j=0;j<ichunk;++j)
-           x[j+st->filt_len-1]=0;
-       }
-
-       speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk);
-     } else {
-       ichunk = 0;
-       ochunk = 0;
-     }
-
-     for (j=0;j<ochunk+omagic;++j)
-#ifdef FIXED_POINT
-        out[j*ostride_save] = ystack[j];
-#else
-        out[j*ostride_save] = WORD2INT(ystack[j]);
-#endif
-
-     ilen -= ichunk;
-     olen -= ochunk;
-     out += (ochunk+omagic) * ostride_save;
-     if (in)
-       in += ichunk * istride_save;
-   }
-   st->out_stride = ostride_save;
-   *in_len -= ilen;
-   *out_len -= olen;
-
-   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
-{
-   spx_uint32_t i;
-   int istride_save, ostride_save;
-   spx_uint32_t bak_out_len = *out_len;
-   spx_uint32_t bak_in_len = *in_len;
-   istride_save = st->in_stride;
-   ostride_save = st->out_stride;
-   st->in_stride = st->out_stride = st->nb_channels;
-   for (i=0;i<st->nb_channels;i++)
-   {
-      *out_len = bak_out_len;
-      *in_len = bak_in_len;
-      if (in != NULL)
-         speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
-      else
-         speex_resampler_process_float(st, i, NULL, in_len, out+i, out_len);
-   }
-   st->in_stride = istride_save;
-   st->out_stride = ostride_save;
-   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
-{
-   spx_uint32_t i;
-   int istride_save, ostride_save;
-   spx_uint32_t bak_out_len = *out_len;
-   spx_uint32_t bak_in_len = *in_len;
-   istride_save = st->in_stride;
-   ostride_save = st->out_stride;
-   st->in_stride = st->out_stride = st->nb_channels;
-   for (i=0;i<st->nb_channels;i++)
-   {
-      *out_len = bak_out_len;
-      *in_len = bak_in_len;
-      if (in != NULL)
-         speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
-      else
-         speex_resampler_process_int(st, i, NULL, in_len, out+i, out_len);
-   }
-   st->in_stride = istride_save;
-   st->out_stride = ostride_save;
-   return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
-   return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
-}
-
-EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
-{
-   *in_rate = st->in_rate;
-   *out_rate = st->out_rate;
-}
-
-EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
-   spx_uint32_t fact;
-   spx_uint32_t old_den;
-   spx_uint32_t i;
-
-   if (ratio_num == 0 || ratio_den == 0)
-      return RESAMPLER_ERR_INVALID_ARG;
-
-   if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den)
-      return RESAMPLER_ERR_SUCCESS;
-
-   old_den = st->den_rate;
-   st->in_rate = in_rate;
-   st->out_rate = out_rate;
-   st->num_rate = ratio_num;
-   st->den_rate = ratio_den;
-   /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
-   for (fact=2;fact<=IMIN(st->num_rate, st->den_rate);fact++)
-   {
-      while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0))
-      {
-         st->num_rate /= fact;
-         st->den_rate /= fact;
-      }
-   }
-
-   if (old_den > 0)
-   {
-      for (i=0;i<st->nb_channels;i++)
-      {
-         st->samp_frac_num[i]=st->samp_frac_num[i]*st->den_rate/old_den;
-         /* Safety net */
-         if (st->samp_frac_num[i] >= st->den_rate)
-            st->samp_frac_num[i] = st->den_rate-1;
-      }
-   }
-
-   if (st->initialised)
-      return update_filter(st);
-   return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
-{
-   *ratio_num = st->num_rate;
-   *ratio_den = st->den_rate;
-}
-
-EXPORT int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
-{
-   if (quality > 10 || quality < 0)
-      return RESAMPLER_ERR_INVALID_ARG;
-   if (st->quality == quality)
-      return RESAMPLER_ERR_SUCCESS;
-   st->quality = quality;
-   if (st->initialised)
-      return update_filter(st);
-   return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
-{
-   *quality = st->quality;
-}
-
-EXPORT void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
-{
-   st->in_stride = stride;
-}
-
-EXPORT void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
-{
-   *stride = st->in_stride;
-}
-
-EXPORT void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
-{
-   st->out_stride = stride;
-}
-
-EXPORT void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
-{
-   *stride = st->out_stride;
-}
-
-EXPORT int speex_resampler_get_input_latency(SpeexResamplerState *st)
-{
-  return st->filt_len / 2;
-}
-
-EXPORT int speex_resampler_get_output_latency(SpeexResamplerState *st)
-{
-  return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) / st->num_rate;
-}
-
-EXPORT int speex_resampler_skip_zeros(SpeexResamplerState *st)
-{
-   spx_uint32_t i;
-   for (i=0;i<st->nb_channels;i++)
-      st->last_sample[i] = st->filt_len/2;
-   return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_reset_mem(SpeexResamplerState *st)
-{
-   spx_uint32_t i;
-   for (i=0;i<st->nb_channels;i++)
-   {
-      st->last_sample[i] = 0;
-      st->magic_samples[i] = 0;
-      st->samp_frac_num[i] = 0;
-   }
-   for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
-      st->mem[i] = 0;
-   return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT const char *speex_resampler_strerror(int err)
-{
-   switch (err)
-   {
-      case RESAMPLER_ERR_SUCCESS:
-         return "Success.";
-      case RESAMPLER_ERR_ALLOC_FAILED:
-         return "Memory allocation failed.";
-      case RESAMPLER_ERR_BAD_STATE:
-         return "Bad resampler state.";
-      case RESAMPLER_ERR_INVALID_ARG:
-         return "Invalid argument.";
-      case RESAMPLER_ERR_PTR_OVERLAP:
-         return "Input and output buffers overlap.";
-      default:
-         return "Unknown error. Bad error code or strange version mismatch.";
-   }
-}
diff --git a/speexdsp/speex_resampler.h b/speexdsp/speex_resampler.h
deleted file mode 100644
index 284e7ec8ed..0000000000
--- a/speexdsp/speex_resampler.h
+++ /dev/null
@@ -1,347 +0,0 @@
-/** @file
-
-   Copyright (C) 2007 Jean-Marc Valin
-
-   File: speex_resampler.h
-   Resampling code
-
-   The design goals of this code are:
-      - Very fast algorithm
-      - Low memory requirement
-      - Good *perceptual* quality (and not best SNR)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions are
-   met:
-
-   1. Redistributions of source code must retain the above copyright notice,
-   this list of conditions and the following disclaimer.
-
-   2. Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   3. The name of the author may not be used to endorse or promote products
-   derived from this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
-   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
-   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
-   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
-   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
-   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
-   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
-   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-   POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef SPEEX_RESAMPLER_H
-#define SPEEX_RESAMPLER_H
-
-#define OUTSIDE_SPEEX 1
-#define RANDOM_PREFIX ws_codec
-#include <ws_symbol_export.h>
-#include <ws_diag_control.h>
-#define EXPORT
-
-#ifdef OUTSIDE_SPEEX
-
-/********* WARNING: MENTAL SANITY ENDS HERE *************/
-
-/* If the resampler is defined outside of Speex, we change the symbol names so that
-   there won't be any clash if linking with Speex later on. */
-
-/* #define RANDOM_PREFIX your software name here */
-#ifndef RANDOM_PREFIX
-#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
-#endif
-
-#define CAT_PREFIX2(a,b) a ## b
-#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
-
-#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
-#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
-#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
-#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
-#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
-#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
-#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
-#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
-#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
-#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
-#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
-#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
-#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
-#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
-#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
-#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride)
-#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
-#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
-#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
-#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
-#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
-#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
-
-#define spx_int16_t short
-#define spx_int32_t int
-#define spx_uint16_t unsigned short
-#define spx_uint32_t unsigned int
-
-#else /* OUTSIDE_SPEEX */
-
-#include "speexdsp_types.h"
-
-#endif /* OUTSIDE_SPEEX */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define SPEEX_RESAMPLER_QUALITY_MAX 10
-#define SPEEX_RESAMPLER_QUALITY_MIN 0
-#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
-#define SPEEX_RESAMPLER_QUALITY_VOIP 3
-#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
-
-enum {
-   RESAMPLER_ERR_SUCCESS         = 0,
-   RESAMPLER_ERR_ALLOC_FAILED    = 1,
-   RESAMPLER_ERR_BAD_STATE       = 2,
-   RESAMPLER_ERR_INVALID_ARG     = 3,
-   RESAMPLER_ERR_PTR_OVERLAP     = 4,
-
-   RESAMPLER_ERR_MAX_ERROR
-};
-
-struct SpeexResamplerState_;
-typedef struct SpeexResamplerState_ SpeexResamplerState;
-
-/** Create a new resampler with integer input and output rates.
- * @param nb_channels Number of channels to be processed
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- */
-SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
-                                                        spx_uint32_t in_rate,
-                                                        spx_uint32_t out_rate,
-                                                        int quality,
-                                                        int *err);
-
-/** Create a new resampler with fractional input/output rates. The sampling
- * rate ratio is an arbitrary rational number with both the numerator and
- * denominator being 32-bit integers.
- * @param nb_channels Number of channels to be processed
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- */
-SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
-                                                             spx_uint32_t ratio_num,
-                                                             spx_uint32_t ratio_den,
-                                                             spx_uint32_t in_rate,
-                                                             spx_uint32_t out_rate,
-                                                             int quality,
-                                                             int *err);
-
-/** Destroy a resampler state.
- * @param st Resampler state
- */
-void speex_resampler_destroy(SpeexResamplerState *st);
-
-/** Resample a float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the
- * number of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_float(SpeexResamplerState *st,
-                                                spx_uint32_t channel_index,
-                                                const float *in,
-                                                spx_uint32_t *in_len,
-                                                float *out,
-                                                spx_uint32_t *out_len);
-
-/** Resample an int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_int(SpeexResamplerState *st,
-                                              spx_uint32_t channel_index,
-                                              const spx_int16_t *in,
-                                              spx_uint32_t *in_len,
-                                              spx_int16_t *out,
-                                              spx_uint32_t *out_len);
-
-/** Resample an interleaved float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
-                                                            const float *in,
-                                                            spx_uint32_t *in_len,
-                                                            float *out,
-                                                            spx_uint32_t *out_len);
-
-/** Resample an interleaved int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
-                                                          const spx_int16_t *in,
-                                                          spx_uint32_t *in_len,
-                                                          spx_int16_t *out,
-                                                          spx_uint32_t *out_len);
-
-/** Set (change) the input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- */
-int speex_resampler_set_rate(SpeexResamplerState *st,
-                                           spx_uint32_t in_rate,
-                                           spx_uint32_t out_rate);
-
-/** Get the current input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz) copied.
- * @param out_rate Output sampling rate (integer number of Hz) copied.
- */
-void speex_resampler_get_rate(SpeexResamplerState *st,
-                                            spx_uint32_t *in_rate,
-                                            spx_uint32_t *out_rate);
-
-/** Set (change) the input/output sampling rates and resampling ratio
- * (fractional values in Hz supported).
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- */
-int speex_resampler_set_rate_frac(SpeexResamplerState *st,
-                                                spx_uint32_t ratio_num,
-                                                spx_uint32_t ratio_den,
-                                                spx_uint32_t in_rate,
-                                                spx_uint32_t out_rate);
-
-/** Get the current resampling ratio. This will be reduced to the least
- * common denominator.
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio copied
- * @param ratio_den Denominator of the sampling rate ratio copied
- */
-void speex_resampler_get_ratio(SpeexResamplerState *st,
-                                             spx_uint32_t *ratio_num,
-                                             spx_uint32_t *ratio_den);
-
-/** Set (change) the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-int speex_resampler_set_quality(SpeexResamplerState *st,
-                                              int quality);
-
-/** Get the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-void speex_resampler_get_quality(SpeexResamplerState *st,
-                                               int *quality);
-
-/** Set (change) the input stride.
- * @param st Resampler state
- * @param stride Input stride
- */
-void speex_resampler_set_input_stride(SpeexResamplerState *st,
-                                                    spx_uint32_t stride);
-
-/** Get the input stride.
- * @param st Resampler state
- * @param stride Input stride copied
- */
-void speex_resampler_get_input_stride(SpeexResamplerState *st,
-                                                    spx_uint32_t *stride);
-
-/** Set (change) the output stride.
- * @param st Resampler state
- * @param stride Output stride
- */
-void speex_resampler_set_output_stride(SpeexResamplerState *st,
-                                                     spx_uint32_t stride);
-
-/** Get the output stride.
- * @param st Resampler state copied
- * @param stride Output stride
- */
-void speex_resampler_get_output_stride(SpeexResamplerState *st,
-                                                     spx_uint32_t *stride);
-
-/** Get the latency introduced by the resampler measured in input samples.
- * @param st Resampler state
- */
-int speex_resampler_get_input_latency(SpeexResamplerState *st);
-
-/** Get the latency introduced by the resampler measured in output samples.
- * @param st Resampler state
- */
-int speex_resampler_get_output_latency(SpeexResamplerState *st);
-
-/** Make sure that the first samples to go out of the resamplers don't have
- * leading zeros. This is only useful before starting to use a newly created
- * resampler. It is recommended to use that when resampling an audio file, as
- * it will generate a file with the same length. For real-time processing,
- * it is probably easier not to use this call (so that the output duration
- * is the same for the first frame).
- * @param st Resampler state
- */
-int speex_resampler_skip_zeros(SpeexResamplerState *st);
-
-/** Reset a resampler so a new (unrelated) stream can be processed.
- * @param st Resampler state
- */
-int speex_resampler_reset_mem(SpeexResamplerState *st);
-
-/** Returns the English meaning for an error code
- * @param err Error code
- * @return English string
- */
-const char *speex_resampler_strerror(int err);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
diff --git a/speexdsp/stack_alloc.h b/speexdsp/stack_alloc.h
deleted file mode 100644
index a446065825..0000000000
--- a/speexdsp/stack_alloc.h
+++ /dev/null
@@ -1,115 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin */
-/**
-   @file stack_alloc.h
-   @brief Temporary memory allocation on stack
-*/
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef STACK_ALLOC_H
-#define STACK_ALLOC_H
-
-#ifdef USE_ALLOCA
-# ifdef _WIN32
-#  include <malloc.h>
-# else
-#  ifdef HAVE_ALLOCA_H
-#   include <alloca.h>
-#  else
-#   include <stdlib.h>
-#  endif
-# endif
-#endif
-
-/**
- * @def ALIGN(stack, size)
- *
- * Aligns the stack to a 'size' boundary
- *
- * @param stack Stack
- * @param size  New size boundary
- */
-
-/**
- * @def PUSH(stack, size, type)
- *
- * Allocates 'size' elements of type 'type' on the stack
- *
- * @param stack Stack
- * @param size  Number of elements
- * @param type  Type of element
- */
-
-/**
- * @def VARDECL(var)
- *
- * Declare variable on stack
- *
- * @param var Variable to declare
- */
-
-/**
- * @def ALLOC(var, size, type)
- *
- * Allocate 'size' elements of 'type' on stack
- *
- * @param var  Name of variable to allocate
- * @param size Number of elements
- * @param type Type of element
- */
-
-#ifdef ENABLE_VALGRIND
-
-#include <valgrind/memcheck.h>
-
-#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1))
-
-#define PUSH(stack, size, type) (VALGRIND_MAKE_NOACCESS(stack, 1000),ALIGN((stack),sizeof(type)),VALGRIND_MAKE_WRITABLE(stack, ((size)*sizeof(type))),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type))))
-
-#else
-
-#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1))
-
-#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type))))
-
-#endif
-
-#if defined(VAR_ARRAYS)
-#define VARDECL(var)
-#define ALLOC(var, size, type) type var[size]
-#elif defined(USE_ALLOCA)
-#define VARDECL(var) var
-#define ALLOC(var, size, type) var = alloca(sizeof(type)*(size))
-#else
-#define VARDECL(var) var
-#define ALLOC(var, size, type) var = PUSH(stack, size, type)
-#endif
-
-
-#endif
diff --git a/tools/win-setup.ps1 b/tools/win-setup.ps1
index 41534b8952..b4d19cb6c3 100644
--- a/tools/win-setup.ps1
+++ b/tools/win-setup.ps1
@@ -92,6 +92,7 @@ $Win64Archives = @{
     "sbc/sbc-1.3-1-win64ws.zip" = "08cef6898c421277a6582ef3225d8820f74a037cbd5b6e673a4d8f4593ce80a1";
     "snappy/snappy-1.1.9-1-win64ws.zip" = "fa907724be019bcc55d27ebe88257ba8898b5c38b719099b8164ac78600d81cc";
     "spandsp/spandsp-0.0.6-2-win64ws.zip" = "2eb8278633037f60f44815ea1606486ab5dcdf3bddc500b20c9fe356856236b2";
+    "speexdsp/speexdsp-1.21.1-1-win64ws.zip" = "d36db62e64ffaee38d9f607bef07d3778d8957ad29757f3eba169eb135f1a4e5";
     "vcpkg-export/vcpkg-export-20220726-1-win64ws.zip" = "b1eaa8124802532fa8d30789219906f90fb80908844e4458327b3f73995a44b0";
     "WinSparkle/WinSparkle-0.5.7.zip" = "56d396ef0c4e8b0589ea74134e484376ca6459d972cd1ab1da6b9624d82e6d04";
     "zstd/zstd-1.5.2-1-win64ws.zip" = "d920afe636951cfcf144824d9c075d1f2c13387f4739152fe185fd9c09fc58f2";
@@ -141,6 +142,7 @@ $CleanupItems = @(
     "sbc-1.3-win??ws"
     "snappy-1.1.*-win??ws"
     "spandsp-0.0.6-win??ws"
+    "speexdsp-*-win??ws"
     "user-guide"
     "vcpkg-export-*-win??ws"
     "zstd-*-win??ws"
diff --git a/ui/logray/logray_main.cpp b/ui/logray/logray_main.cpp
index 774840593c..f1e67b30b0 100644
--- a/ui/logray/logray_main.cpp
+++ b/ui/logray/logray_main.cpp
@@ -225,11 +225,6 @@ gather_wireshark_qt_compiled_info(feature_list l)
     without_feature(l, "AirPcap");
 #endif
 #endif /* _WIN32 */
-#ifdef HAVE_SPEEXDSP
-    with_feature(l, "SpeexDSP (using system library)");
-#else
-    with_feature(l, "SpeexDSP (using bundled resampler)");
-#endif
 
 #ifdef HAVE_MINIZIP
     with_feature(l, "Minizip");
diff --git a/ui/qt/main.cpp b/ui/qt/main.cpp
index 23da6a4083..68b53ffa7e 100644
--- a/ui/qt/main.cpp
+++ b/ui/qt/main.cpp
@@ -227,11 +227,6 @@ gather_wireshark_qt_compiled_info(feature_list l)
     without_feature(l, "AirPcap");
 #endif
 #endif /* _WIN32 */
-#ifdef HAVE_SPEEXDSP
-    with_feature(l, "SpeexDSP (using system library)");
-#else
-    with_feature(l, "SpeexDSP (using bundled resampler)");
-#endif
 
 #ifdef HAVE_MINIZIP
     with_feature(l, "Minizip");
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 4229854c07..6f93c12403 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -11,11 +11,7 @@
 
 #ifdef QT_MULTIMEDIA_LIB
 
-#ifdef HAVE_SPEEXDSP
 #include <speex/speex_resampler.h>
-#else
-#include "../../speexdsp/speex_resampler.h"
-#endif /* HAVE_SPEEXDSP */
 
 #include <epan/rtp_pt.h>
 #include <epan/to_str.h>
diff --git a/ui/qt/utils/rtp_audio_file.h b/ui/qt/utils/rtp_audio_file.h
index 3024d29287..addc3015d5 100644
--- a/ui/qt/utils/rtp_audio_file.h
+++ b/ui/qt/utils/rtp_audio_file.h
@@ -13,11 +13,7 @@
 #include "config.h"
 #include <ui/rtp_media.h>
 
-#ifdef HAVE_SPEEXDSP
 #include <speex/speex_resampler.h>
-#else
-#include "../../speexdsp/speex_resampler.h"
-#endif /* HAVE_SPEEXDSP */
 
 #include <QIODevice>
 #include <QDir>
author	João Valverde <j@v6e.pt>	2022-12-15 18:54:48 +0000
committer	João Valverde <j@v6e.pt>	2022-12-16 11:11:28 +0000
commit	ae148498649e38105607e9da2262fc276ffe2049 (patch)
tree	a603adaab2f0b334fb39eb72f0644a7ab91da838
parent	a4f56e90a66e81f85d8e5f81eb3e5781198311c3 (diff)