Qt: Initial RTP playback.

Note the "initial". This is woefully incomplete.  See the "to do" lists
below and in the code.

This differs a bit from the GTK+ version in that you specify one or more
streams to be decoded.

Instead of showing waveforms in individual widgets, add them all to a
single QCustomPlot. This conserves screen real estate and lets us more
easily take advantage of the QCP API. It also looks better IMHO.

Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We
probably won't use the widgets until we make 5.0 our minimum Qt
version and plain old QtMultimedia lets us support Qt 4 more easily
(in theory at least).

Add resampling code from libspeex. I initially used this to resample
each packet to match the preferred rate of our output device, but this
resulted in poorer audio quality than expected. Leave it in and use to
create visual samples for QCP and to match rates any time the rate
changes. The latter is currently untested.

Add some debugging macros.

Note that both the RTP player and RTP analysis dialogs decode audio data
using different code.

Note that voip_calls_packet and voip_calls_init_tap appear to be dead
code.

To do:

- Add silence frames where needed.
- Implement the jitter buffer.
- Implement the playback timing controls.
- Tapping / scanning streams might be too slow.

Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4
Bug: 9007
Reviewed-on: https://code.wireshark.org/review/10458
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>

8 files changed, 1923 insertions, 2 deletions

diff --git a/codecs/CMakeLists.txt b/codecs/CMakeLists.txt
index 6397db18c0..63ac64d08d 100644
--- a/codecs/CMakeLists.txt
+++ b/codecs/CMakeLists.txt
@@ -27,6 +27,17 @@ set(CODECS_FILES
   # handle or define.
   # G722/G722decode.c
   # G726/G726decode.c
+  speex/resample.c
+)
+
+# Enables visibility in IDEs
+file(GLOB EXTRA_CODEC_HEADERS
+  codecs.h
+  G711a/G711adecode.h G711a/G711atable.h
+  G711u/G711udecode.h G711u/G711utable.h
+  speex/arch.h
+  speex/speex_resampler.h
+  speex/stack_alloc.h
 )
 
 if(SBC_FOUND)
diff --git a/codecs/Makefile.common b/codecs/Makefile.common
index fc76046c34..d076d3b412 100644
--- a/codecs/Makefile.common
+++ b/codecs/Makefile.common
@@ -27,7 +27,8 @@ LIBCODEC_SRC = \
 	G711u/G711udecode.c \
 	G722/G722decode.c \
 	G726/G726decode.c \
-	sbc/sbc.c
+	sbc/sbc.c \
+	speex/resample.c
 
 noinst_HEADERS = \
 	codecs.h \
@@ -35,5 +36,8 @@ noinst_HEADERS = \
 	G711u/G711udecode.h G711u/G711utable.h	\
 	G722/G722decode.h \
 	G726/G726decode.h \
-	sbc/sbc_private.h
+	sbc/sbc_private.h \
+	speex/arch.h \
+	speex/speex_resampler.h \
+	speex/stack_alloc.h
 
diff --git a/codecs/Makefile.nmake b/codecs/Makefile.nmake
index 1a8dc6e5cb..53aa69bda3 100644
--- a/codecs/Makefile.nmake
+++ b/codecs/Makefile.nmake
@@ -10,6 +10,9 @@ include ..\Makefile.nmake.inc
 CFLAGS=/I.. $(WARNINGS_ARE_ERRORS) $(STANDARD_CFLAGS) \
 	$(GLIB_CFLAGS)
 
+DIRTY_CFLAGS=/I.. $(STANDARD_CFLAGS) \
+	$(GLIB_CFLAGS)
+
 .c.obj::
 	$(CC) $(CFLAGS) -Fd.\ -c $<
 
@@ -26,6 +29,7 @@ LIBCODEC_OBJECTS= \
 	G711adecode.obj \
 	G722decode.obj \
 	G726decode.obj \
+	resample.obj \
 	sbc.obj
 
 codecs.lib	:  $(LIBCODEC_OBJECTS)
@@ -46,6 +50,9 @@ G722decode.obj: G722\G722decode.c G722\G722decode.h
 G726decode.obj: G726\G726decode.c G726\G726decode.h
 	$(CC) $(CFLAGS) -Fd.\ -c G726\G726decode.c /Fo%|fF.obj
 
+resample.obj: speex\resample.c speex\arch.h speex\speex_resampler.h speex\stack_alloc.h
+	$(CC) $(DIRTY_CFLAGS) -Fd.\ -c speex\resample.c /Fo%|fF.obj
+
 sbc.obj: sbc\sbc.c sbc\sbc_private.h
 	$(CC) $(CFLAGS) -Fd.\ -c sbc\sbc.c /Fo%|fF.obj
 
diff --git a/codecs/speex/README.txt b/codecs/speex/README.txt
new file mode 100644
index 0000000000..ee8cd66e13
--- /dev/null
+++ b/codecs/speex/README.txt
@@ -0,0 +1,2 @@
+Copied from http://git.xiph.org/speexdsp.git c470e2e89a6ca75b507437467692cd684b71a526
+and modified to compile in the Wireshark source tree.
diff --git a/codecs/speex/arch.h b/codecs/speex/arch.h
new file mode 100644
index 0000000000..c2de991dd9
--- /dev/null
+++ b/codecs/speex/arch.h
@@ -0,0 +1,235 @@
+/* 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
+
+#define ABS(x) ((x) < 0 ? (-(x)) : (x))      /**< Absolute integer value. */
+#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) ((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) (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)     ((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) ((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/codecs/speex/resample.c b/codecs/speex/resample.c
new file mode 100644
index 0000000000..c5f663432b
--- /dev/null
+++ b/codecs/speex/resample.c
@@ -0,0 +1,1203 @@
+/* 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 <stdlib.h>
+static void *speex_alloc (int size) {return calloc(size,1);}
+static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
+static void speex_free (void *ptr) {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>
+
+#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
+
+#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 cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(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] = 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 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/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/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/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 (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 (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/codecs/speex/speex_resampler.h b/codecs/speex/speex_resampler.h
new file mode 100644
index 0000000000..a71cd4fb68
--- /dev/null
+++ b/codecs/speex/speex_resampler.h
@@ -0,0 +1,344 @@
+/* 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"
+#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/codecs/speex/stack_alloc.h b/codecs/speex/stack_alloc.h
new file mode 100644
index 0000000000..6c56334f86
--- /dev/null
+++ b/codecs/speex/stack_alloc.h
@@ -0,0 +1,115 @@
+/* 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