Add resample element based on the Speex resampling algorithm.

Original commit message from CVS:
* configure.ac:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
(gst_speex_resample_init), (gst_speex_resample_start),
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
(gst_speex_resample_transform_caps),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_event), (gst_speex_resample_check_discont),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_set_property),
(gst_speex_resample_get_property), (plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
(resampler_basic_direct_single), (resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init), (speex_resampler_init_frac),
(speex_resampler_destroy), (speex_resampler_process_native),
(speex_resampler_process_float), (speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate), (speex_resampler_get_rate),
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
(speex_resampler_set_quality), (speex_resampler_get_quality),
(speex_resampler_set_input_stride),
(speex_resampler_get_input_stride),
(speex_resampler_set_output_stride),
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem), (speex_resampler_strerror):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add resample element based on the Speex resampling algorithm.

11 files changed, 2969 insertions, 0 deletions

diff --git a/ChangeLog b/ChangeLog
index 56b8d3dc..a77f8c7c 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,45 @@
+2007-11-20  Sebastian Dröge  <slomo@circular-chaos.org>
+
+	* configure.ac:
+	* gst/speexresample/arch.h:
+	* gst/speexresample/fixed_generic.h:
+	* gst/speexresample/gstspeexresample.c:
+	(gst_speex_resample_base_init), (gst_speex_resample_class_init),
+	(gst_speex_resample_init), (gst_speex_resample_start),
+	(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
+	(gst_speex_resample_transform_caps),
+	(gst_speex_resample_init_state), (gst_speex_resample_update_state),
+	(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
+	(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
+	(gst_speex_resample_event), (gst_speex_resample_check_discont),
+	(gst_speex_resample_process), (gst_speex_resample_transform),
+	(gst_speex_resample_set_property),
+	(gst_speex_resample_get_property), (plugin_init):
+	* gst/speexresample/gstspeexresample.h:
+	* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
+	(speex_free), (compute_func), (main), (sinc), (cubic_coef),
+	(resampler_basic_direct_single), (resampler_basic_direct_double),
+	(resampler_basic_interpolate_single),
+	(resampler_basic_interpolate_double), (update_filter),
+	(speex_resampler_init), (speex_resampler_init_frac),
+	(speex_resampler_destroy), (speex_resampler_process_native),
+	(speex_resampler_process_float), (speex_resampler_process_int),
+	(speex_resampler_process_interleaved_float),
+	(speex_resampler_process_interleaved_int),
+	(speex_resampler_set_rate), (speex_resampler_get_rate),
+	(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
+	(speex_resampler_set_quality), (speex_resampler_get_quality),
+	(speex_resampler_set_input_stride),
+	(speex_resampler_get_input_stride),
+	(speex_resampler_set_output_stride),
+	(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
+	(speex_resampler_reset_mem), (speex_resampler_strerror):
+	* gst/speexresample/speex_resampler.h:
+	* gst/speexresample/speex_resampler_float.c:
+	* gst/speexresample/speex_resampler_int.c:
+	* gst/speexresample/speex_resampler_wrapper.h:
+	Add resample element based on the Speex resampling algorithm.
+
 2007-11-20  Edgard Lima  <edgard.lima@indt.org.br>
 
 	* ext/metadata/gstmetadataparse.c:
diff --git a/configure.ac b/configure.ac
index b6ab0cf7..dcf4b35d 100644
--- a/configure.ac
+++ b/configure.ac
@@ -105,6 +105,7 @@ GST_PLUGINS_ALL="\
   rtpmanager \
   sdp \
   spectrum \
+  speexresample \
   speed \
   stereo \
   switch \
@@ -1069,6 +1070,7 @@ gst/rtpmanager/Makefile
 gst/sdp/Makefile
 gst/spectrum/Makefile
 gst/speed/Makefile
+gst/speexresample/Makefile
 gst/stereo/Makefile
 gst/switch/Makefile
 gst/tta/Makefile
diff --git a/gst/speexresample/arch.h b/gst/speexresample/arch.h
new file mode 100644
index 00000000..f213e683
--- /dev/null
+++ b/gst/speexresample/arch.h
@@ -0,0 +1,243 @@
+/* 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
+
+#ifndef SPEEX_VERSION
+#define SPEEX_MAJOR_VERSION 1         /**< Major Speex version. */
+#define SPEEX_MINOR_VERSION 1         /**< Minor Speex version. */
+#define SPEEX_MICRO_VERSION 15        /**< Micro Speex version. */
+#define SPEEX_EXTRA_VERSION ""        /**< Extra Speex version. */
+#define SPEEX_VERSION "speex-1.2beta3"  /**< Speex version string. */
+#endif
+
+/* 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/speex_types.h"
+#endif
+
+#ifndef ABS
+#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 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 (ARM5E_ASM)
+#include "fixed_arm5e.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 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))
+
+
+#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
+long long spx_mips=0;
+#endif
+
+
+#endif
diff --git a/gst/speexresample/fixed_generic.h b/gst/speexresample/fixed_generic.h
new file mode 100644
index 00000000..2948177c
--- /dev/null
+++ b/gst/speexresample/fixed_generic.h
@@ -0,0 +1,106 @@
+/* Copyright (C) 2003 Jean-Marc Valin */
+/**
+   @file fixed_generic.h
+   @brief Generic fixed-point operations
+*/
+/*
+   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 FIXED_GENERIC_H
+#define FIXED_GENERIC_H
+
+#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
+#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
+
+#define NEG16(x) (-(x))
+#define NEG32(x) (-(x))
+#define EXTRACT16(x) ((spx_word16_t)(x))
+#define EXTEND32(x) ((spx_word32_t)(x))
+#define SHR16(a,shift) ((a) >> (shift))
+#define SHL16(a,shift) ((a) << (shift))
+#define SHR32(a,shift) ((a) >> (shift))
+#define SHL32(a,shift) ((a) << (shift))
+#define PSHR16(a,shift) (SHR16((a)+((1<<((shift))>>1)),shift))
+#define PSHR32(a,shift) (SHR32((a)+((1<<((shift))>>1)),shift))
+#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift)))
+#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
+#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
+
+#define SHR(a,shift) ((a) >> (shift))
+#define SHL(a,shift) ((spx_word32_t)(a) << (shift))
+#define PSHR(a,shift) (SHR((a)+((1<<((shift))>>1)),shift))
+#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
+
+
+#define ADD16(a,b) ((spx_word16_t)((spx_word16_t)(a)+(spx_word16_t)(b)))
+#define SUB16(a,b) ((spx_word16_t)(a)-(spx_word16_t)(b))
+#define ADD32(a,b) ((spx_word32_t)(a)+(spx_word32_t)(b))
+#define SUB32(a,b) ((spx_word32_t)(a)-(spx_word32_t)(b))
+
+
+/* result fits in 16 bits */
+#define MULT16_16_16(a,b)     ((((spx_word16_t)(a))*((spx_word16_t)(b))))
+
+/* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */
+#define MULT16_16(a,b)     (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b)))
+
+#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b))))
+#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12))
+#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13))
+#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14))
+
+#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))
+#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)))
+
+#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15))
+#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))
+#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)))
+
+
+#define MAC16_16_Q11(c,a,b)     (ADD32((c),SHR(MULT16_16((a),(b)),11)))
+#define MAC16_16_Q13(c,a,b)     (ADD32((c),SHR(MULT16_16((a),(b)),13)))
+#define MAC16_16_P13(c,a,b)     (ADD32((c),SHR(ADD32(4096,MULT16_16((a),(b))),13)))
+
+#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11))
+#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13))
+#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14))
+#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15))
+
+#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13))
+#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14))
+#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15))
+
+#define MUL_16_32_R15(a,bh,bl) ADD32(MULT16_16((a),(bh)), SHR(MULT16_16((a),(bl)),15))
+
+#define DIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a))/((spx_word16_t)(b))))
+#define PDIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((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_word16_t)(b)>>1))/((spx_word32_t)(b)))
+
+#endif
diff --git a/gst/speexresample/gstspeexresample.c b/gst/speexresample/gstspeexresample.c
new file mode 100644
index 00000000..307b8180
--- /dev/null
+++ b/gst/speexresample/gstspeexresample.c
@@ -0,0 +1,733 @@
+/* GStreamer
+ * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-speexresample
+ *
+ * <refsect2>
+ * speexresample resamples raw audio buffers to different sample rates using
+ * a configurable windowing function to enhance quality.
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! speexresample ! audio/x-raw-int, rate=8000 ! alsasink
+ * </programlisting>
+ * Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa. 
+ * To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
+ * </para>
+ * </refsect2>
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <math.h>
+
+#include "gstspeexresample.h"
+#include <gst/audio/audio.h>
+#include <gst/base/gstbasetransform.h>
+
+GST_DEBUG_CATEGORY (speex_resample_debug);
+#define GST_CAT_DEFAULT speex_resample_debug
+
+enum
+{
+  PROP_0,
+  PROP_QUALITY
+};
+
+#define SUPPORTED_CAPS \
+GST_STATIC_CAPS ( \
+    "audio/x-raw-float, " \
+      "rate = (int) [ 1, MAX ], "	\
+      "channels = (int) [ 1, MAX ], " \
+      "endianness = (int) BYTE_ORDER, " \
+      "width = (int) 32; " \
+    "audio/x-raw-int, " \
+      "rate = (int) [ 1, MAX ], " \
+      "channels = (int) [ 1, MAX ], " \
+      "endianness = (int) BYTE_ORDER, " \
+      "width = (int) 16, " \
+      "depth = (int) 16, " \
+      "signed = (boolean) true" \
+)
+
+static GstStaticPadTemplate gst_speex_resample_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
+
+static GstStaticPadTemplate gst_speex_resample_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
+
+static void gst_speex_resample_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_speex_resample_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+/* vmethods */
+static gboolean gst_speex_resample_get_unit_size (GstBaseTransform * base,
+    GstCaps * caps, guint * size);
+static GstCaps *gst_speex_resample_transform_caps (GstBaseTransform * base,
+    GstPadDirection direction, GstCaps * caps);
+static gboolean gst_speex_resample_transform_size (GstBaseTransform * trans,
+    GstPadDirection direction, GstCaps * incaps, guint insize,
+    GstCaps * outcaps, guint * outsize);
+static gboolean gst_speex_resample_set_caps (GstBaseTransform * base,
+    GstCaps * incaps, GstCaps * outcaps);
+static GstFlowReturn gst_speex_resample_transform (GstBaseTransform * base,
+    GstBuffer * inbuf, GstBuffer * outbuf);
+static gboolean gst_speex_resample_event (GstBaseTransform * base,
+    GstEvent * event);
+static gboolean gst_speex_resample_start (GstBaseTransform * base);
+static gboolean gst_speex_resample_stop (GstBaseTransform * base);
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (speex_resample_debug, "speex_resample", 0, "audio resampling element");
+
+GST_BOILERPLATE_FULL (GstSpeexResample, gst_speex_resample, GstBaseTransform,
+    GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
+
+static void
+gst_speex_resample_base_init (gpointer g_class)
+{
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_speex_resample_src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_speex_resample_sink_template));
+
+  gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
+      "Filter/Converter/Audio", "Resamples audio",
+      "Sebastian Dröge <slomo@circular-chaos.org>");
+}
+
+static void
+gst_speex_resample_class_init (GstSpeexResampleClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  gobject_class->set_property = gst_speex_resample_set_property;
+  gobject_class->get_property = gst_speex_resample_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_QUALITY,
+      g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
+          "the lowest and 10 being the best",
+          SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
+          SPEEX_RESAMPLER_QUALITY_DEFAULT,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+
+  GST_BASE_TRANSFORM_CLASS (klass)->start =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_start);
+  GST_BASE_TRANSFORM_CLASS (klass)->stop =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_stop);
+  GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_transform_size);
+  GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_get_unit_size);
+  GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_transform_caps);
+  GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_set_caps);
+  GST_BASE_TRANSFORM_CLASS (klass)->transform =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_transform);
+  GST_BASE_TRANSFORM_CLASS (klass)->event =
+      GST_DEBUG_FUNCPTR (gst_speex_resample_event);
+
+  GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
+}
+
+static void
+gst_speex_resample_init (GstSpeexResample * resample,
+    GstSpeexResampleClass * klass)
+{
+  resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
+
+  resample->need_discont = FALSE;
+}
+
+/* vmethods */
+static gboolean
+gst_speex_resample_start (GstBaseTransform * base)
+{
+  GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
+
+  resample->ts_offset = -1;
+  resample->offset = -1;
+  resample->next_ts = -1;
+
+  return TRUE;
+}
+
+static gboolean
+gst_speex_resample_stop (GstBaseTransform * base)
+{
+  GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
+
+  if (resample->state) {
+    resample_resampler_destroy (resample->state);
+    resample->state = NULL;
+  }
+
+  gst_caps_replace (&resample->sinkcaps, NULL);
+  gst_caps_replace (&resample->srccaps, NULL);
+
+  return TRUE;
+}
+
+static gboolean
+gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
+    guint * size)
+{
+  gint width, channels;
+  GstStructure *structure;
+  gboolean ret;
+
+  g_return_val_if_fail (size != NULL, FALSE);
+
+  /* this works for both float and int */
+  structure = gst_caps_get_structure (caps, 0);
+  ret = gst_structure_get_int (structure, "width", &width);
+  ret &= gst_structure_get_int (structure, "channels", &channels);
+  g_return_val_if_fail (ret, FALSE);
+
+  *size = width * channels / 8;
+
+  return TRUE;
+}
+
+static GstCaps *
+gst_speex_resample_transform_caps (GstBaseTransform * base,
+    GstPadDirection direction, GstCaps * caps)
+{
+  GstCaps *res;
+  GstStructure *structure;
+
+  /* transform caps gives one single caps so we can just replace
+   * the rate property with our range. */
+  res = gst_caps_copy (caps);
+  structure = gst_caps_get_structure (res, 0);
+  gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+
+  return res;
+}
+
+static SpeexResamplerState *
+gst_speex_resample_init_state (guint channels, guint inrate, guint outrate,
+    guint quality, gboolean fp)
+{
+  SpeexResamplerState *ret = NULL;
+  gint err = RESAMPLER_ERR_SUCCESS;
+
+  if (fp)
+    ret =
+        resample_float_resampler_init (channels, inrate, outrate, quality,
+        &err);
+  else
+    ret =
+        resample_int_resampler_init (channels, inrate, outrate, quality, &err);
+
+  if (err != RESAMPLER_ERR_SUCCESS) {
+    GST_ERROR ("Failed to create resampler state: %s",
+        resample_resampler_strerror (err));
+    return NULL;
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_speex_resample_update_state (GstSpeexResample * resample, gint channels,
+    gint inrate, gint outrate, gint quality, gboolean fp)
+{
+  gboolean ret = TRUE;
+
+  if (resample->state == NULL) {
+    ret = TRUE;
+  } else if (resample->channels != channels || fp != resample->fp) {
+    resample_resampler_destroy (resample->state);
+    resample->state =
+        gst_speex_resample_init_state (channels, inrate, outrate, quality, fp);
+
+    ret = (resample->state != NULL);
+  } else if (resample->inrate != inrate || resample->outrate != outrate) {
+    gint err = RESAMPLER_ERR_SUCCESS;
+
+    if (fp)
+      err =
+          resample_float_resampler_set_rate (resample->state, inrate, outrate);
+    else
+      err = resample_int_resampler_set_rate (resample->state, inrate, outrate);
+
+    if (err != RESAMPLER_ERR_SUCCESS)
+      GST_ERROR ("Failed to update rate: %s",
+          resample_resampler_strerror (err));
+
+    ret = (err == RESAMPLER_ERR_SUCCESS);
+  } else if (quality != resample->quality) {
+    gint err = RESAMPLER_ERR_SUCCESS;
+
+    if (fp)
+      err = resample_float_resampler_set_quality (resample->state, quality);
+    else
+      err = resample_int_resampler_set_quality (resample->state, quality);
+
+    if (err != RESAMPLER_ERR_SUCCESS)
+      GST_ERROR ("Failed to update quality: %s",
+          resample_resampler_strerror (err));
+
+    ret = (err == RESAMPLER_ERR_SUCCESS);
+  }
+
+  resample->channels = channels;
+  resample->fp = fp;
+  resample->quality = quality;
+  resample->inrate = inrate;
+  resample->outrate = outrate;
+
+  return ret;
+}
+
+static void
+gst_speex_resample_reset_state (GstSpeexResample * resample)
+{
+  if (resample->state && resample->fp)
+    resample_float_resampler_reset_mem (resample->state);
+  else if (resample->state && !resample->fp)
+    resample_int_resampler_reset_mem (resample->state);
+}
+
+static gboolean
+gst_speex_resample_parse_caps (GstCaps * incaps,
+    GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate,
+    gboolean * fp)
+{
+  GstStructure *structure;
+  gboolean ret;
+  gint myinrate, myoutrate, mychannels;
+  gboolean myfp;
+
+  GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
+      GST_PTR_FORMAT, incaps, outcaps);
+
+  structure = gst_caps_get_structure (incaps, 0);
+
+  if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float"))
+    myfp = TRUE;
+  else
+    myfp = FALSE;
+
+  ret = gst_structure_get_int (structure, "rate", &myinrate);
+  ret &= gst_structure_get_int (structure, "channels", &mychannels);
+  if (!ret)
+    goto no_in_rate_channels;
+
+  structure = gst_caps_get_structure (outcaps, 0);
+  ret = gst_structure_get_int (structure, "rate", &myoutrate);
+  if (!ret)
+    goto no_out_rate;
+
+  if (channels)
+    *channels = mychannels;
+  if (inrate)
+    *inrate = myinrate;
+  if (outrate)
+    *outrate = myoutrate;
+
+  if (fp)
+    *fp = myfp;
+
+  return TRUE;
+
+  /* ERRORS */
+no_in_rate_channels:
+  {
+    GST_DEBUG ("could not get input rate and channels");
+    return FALSE;
+  }
+no_out_rate:
+  {
+    GST_DEBUG ("could not get output rate");
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_speex_resample_transform_size (GstBaseTransform * base,
+    GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
+    guint * othersize)
+{
+  GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
+  SpeexResamplerState *state;
+  GstCaps *srccaps, *sinkcaps;
+  gboolean use_internal = FALSE;        /* whether we use the internal state */
+  gboolean ret = TRUE;
+  guint32 ratio_den, ratio_num;
+  gboolean fp;
+
+  GST_LOG ("asked to transform size %d in direction %s",
+      size, direction == GST_PAD_SINK ? "SINK" : "SRC");
+  if (direction == GST_PAD_SINK) {
+    sinkcaps = caps;
+    srccaps = othercaps;
+  } else {
+    sinkcaps = othercaps;
+    srccaps = caps;
+  }
+
+  /* if the caps are the ones that _set_caps got called with; we can use
+   * our own state; otherwise we'll have to create a state */
+  if (resample->state && gst_caps_is_equal (sinkcaps, resample->sinkcaps) &&
+      gst_caps_is_equal (srccaps, resample->srccaps)) {
+    use_internal = TRUE;
+    state = resample->state;
+    fp = resample->fp;
+  } else {
+    gint inrate, outrate, channels;
+
+    GST_DEBUG ("Can't use internal state, creating state");
+
+    ret =
+        gst_speex_resample_parse_caps (caps, othercaps, &channels, &inrate,
+        &outrate, &fp);
+
+    if (!ret) {
+      GST_ERROR ("Wrong caps");
+      return FALSE;
+    }
+
+    state = gst_speex_resample_init_state (channels, inrate, outrate, 0, TRUE);
+    if (!state)
+      return FALSE;
+  }
+
+  if (resample->fp || use_internal)
+    resample_float_resampler_get_ratio (state, &ratio_num, &ratio_den);
+  else
+    resample_int_resampler_get_ratio (state, &ratio_num, &ratio_den);
+
+  if (direction == GST_PAD_SINK) {
+    gint fac = (fp) ? 4 : 2;
+
+    /* asked to convert size of an incoming buffer */
+    size /= fac;
+    *othersize = (size * ratio_den + (ratio_num >> 1)) / ratio_num;
+    *othersize *= fac;
+  } else {
+    gint fac = (fp) ? 4 : 2;
+
+    /* asked to convert size of an outgoing buffer */
+    size /= fac;
+    *othersize = (size * ratio_num + (ratio_den >> 1)) / ratio_den;
+    *othersize *= fac;
+  }
+
+  GST_LOG ("transformed size %d to %d", size, *othersize);
+
+  if (!use_internal)
+    resample_resampler_destroy (state);
+
+  return ret;
+}
+
+static gboolean
+gst_speex_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+    GstCaps * outcaps)
+{
+  gboolean ret;
+  gint inrate = 0, outrate = 0, channels = 0;
+  gboolean fp = FALSE;
+  GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
+
+  GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
+      GST_PTR_FORMAT, incaps, outcaps);
+
+  ret = gst_speex_resample_parse_caps (incaps, outcaps,
+      &channels, &inrate, &outrate, &fp);
+
+  g_return_val_if_fail (ret, FALSE);
+
+  ret =
+      gst_speex_resample_update_state (resample, channels, inrate, outrate,
+      resample->quality, fp);
+
+  g_return_val_if_fail (ret, FALSE);
+
+  /* save caps so we can short-circuit in the size_transform if the caps
+   * are the same */
+  gst_caps_replace (&resample->sinkcaps, incaps);
+  gst_caps_replace (&resample->srccaps, outcaps);
+
+  return TRUE;
+}
+
+static gboolean
+gst_speex_resample_event (GstBaseTransform * base, GstEvent * event)
+{
+  GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_FLUSH_START:
+      break;
+    case GST_EVENT_FLUSH_STOP:
+    case GST_EVENT_NEWSEGMENT:
+      gst_speex_resample_reset_state (resample);
+      resample->ts_offset = -1;
+      resample->next_ts = -1;
+      resample->offset = -1;
+      break;
+    case GST_EVENT_EOS:
+      gst_speex_resample_reset_state (resample);
+      break;
+    default:
+      break;
+  }
+  parent_class->event (base, event);
+
+  return TRUE;
+}
+
+static gboolean
+gst_speex_resample_check_discont (GstSpeexResample * resample,
+    GstClockTime timestamp)
+{
+  if (timestamp != GST_CLOCK_TIME_NONE &&
+      resample->prev_ts != GST_CLOCK_TIME_NONE &&
+      resample->prev_duration != GST_CLOCK_TIME_NONE &&
+      timestamp != resample->prev_ts + resample->prev_duration) {
+    /* Potentially a discontinuous buffer. However, it turns out that many
+     * elements generate imperfect streams due to rounding errors, so we permit
+     * a small error (up to one sample) without triggering a filter 
+     * flush/restart (if triggered incorrectly, this will be audible) */
+    GstClockTimeDiff diff = timestamp -
+        (resample->prev_ts + resample->prev_duration);
+
+    if (ABS (diff) > GST_SECOND / resample->inrate) {
+      GST_WARNING ("encountered timestamp discontinuity of %" G_GINT64_FORMAT,
+          diff);
+      return TRUE;
+    }
+  }
+
+  return FALSE;
+}
+
+static GstFlowReturn
+gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
+    GstBuffer * outbuf)
+{
+  guint32 in_len, in_processed;
+  guint32 out_len, out_processed;
+  gint err = RESAMPLER_ERR_SUCCESS;
+
+  in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
+  out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
+
+  if (resample->fp) {
+    in_len /= 4;
+    out_len /= 4;
+  } else {
+    in_len /= 2;
+    out_len /= 2;
+  }
+
+  in_processed = in_len;
+  out_processed = out_len;
+
+  if (resample->fp)
+    err = resample_float_resampler_process_interleaved_float (resample->state,
+        (const gfloat *) GST_BUFFER_DATA (inbuf), &in_processed,
+        (gfloat *) GST_BUFFER_DATA (outbuf), &out_processed);
+  else
+    err = resample_int_resampler_process_interleaved_int (resample->state,
+        (const gint16 *) GST_BUFFER_DATA (inbuf), &in_processed,
+        (gint16 *) GST_BUFFER_DATA (outbuf), &out_processed);
+
+  if (in_len != in_processed)
+    GST_WARNING ("Converted %d of %d input samples", in_processed, in_len);
+
+  if (out_len != out_processed)
+    GST_WARNING ("Converted to %d instead of %d output samples", out_processed,
+        out_len);
+
+  if (err != RESAMPLER_ERR_SUCCESS) {
+    GST_ERROR ("Failed to convert data: %s", resample_resampler_strerror (err));
+    return GST_FLOW_ERROR;
+  } else {
+    return GST_FLOW_OK;
+  }
+}
+
+static GstFlowReturn
+gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+    GstBuffer * outbuf)
+{
+  GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
+  guint8 *data;
+  gulong size;
+  GstClockTime timestamp;
+  gint outsamples;
+
+  if (resample->state == NULL)
+    if (!(resample->state = gst_speex_resample_init_state (resample->channels,
+                resample->inrate, resample->outrate, resample->quality,
+                resample->fp)))
+      return GST_FLOW_ERROR;
+
+  data = GST_BUFFER_DATA (inbuf);
+  size = GST_BUFFER_SIZE (inbuf);
+  timestamp = GST_BUFFER_TIMESTAMP (inbuf);
+
+  GST_LOG ("transforming buffer of %ld bytes, ts %"
+      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
+      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
+      size, GST_TIME_ARGS (timestamp),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
+      GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
+
+  /* check for timestamp discontinuities and flush/reset if needed */
+  if (G_UNLIKELY (gst_speex_resample_check_discont (resample, timestamp)
+          || GST_BUFFER_IS_DISCONT (inbuf))) {
+    /* Flush internal samples */
+    gst_speex_resample_reset_state (resample);
+    /* Inform downstream element about discontinuity */
+    resample->need_discont = TRUE;
+    /* We want to recalculate the offset */
+    resample->ts_offset = -1;
+  }
+
+  outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
+  outsamples /= (resample->fp) ? 4 : 2;
+
+  if (resample->ts_offset == -1) {
+    /* if we don't know the initial offset yet, calculate it based on the 
+     * input timestamp. */
+    if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+      GstClockTime stime;
+
+      /* offset used to calculate the timestamps. We use the sample offset for
+       * this to make it more accurate. We want the first buffer to have the
+       * same timestamp as the incoming timestamp. */
+      resample->next_ts = timestamp;
+      resample->ts_offset =
+          gst_util_uint64_scale_int (timestamp, resample->outrate, GST_SECOND);
+      /* offset used to set as the buffer offset, this offset is always
+       * relative to the stream time, note that timestamp is not... */
+      stime = (timestamp - base->segment.start) + base->segment.time;
+      resample->offset =
+          gst_util_uint64_scale_int (stime, resample->outrate, GST_SECOND);
+    }
+  }
+  resample->prev_ts = timestamp;
+  resample->prev_duration = GST_BUFFER_DURATION (inbuf);
+
+  GST_BUFFER_OFFSET (outbuf) = resample->offset;
+  GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
+
+  if (resample->ts_offset != -1) {
+    resample->offset += outsamples;
+    resample->ts_offset += outsamples;
+    resample->next_ts =
+        gst_util_uint64_scale_int (resample->ts_offset, GST_SECOND,
+        resample->outrate);
+    GST_BUFFER_OFFSET_END (outbuf) = resample->offset;
+
+    /* we calculate DURATION as the difference between "next" timestamp
+     * and current timestamp so we ensure a contiguous stream, instead of
+     * having rounding errors. */
+    GST_BUFFER_DURATION (outbuf) = resample->next_ts -
+        GST_BUFFER_TIMESTAMP (outbuf);
+  } else {
+    /* no valid offset know, we can still sortof calculate the duration though */
+    GST_BUFFER_DURATION (outbuf) =
+        gst_util_uint64_scale_int (outsamples, GST_SECOND, resample->outrate);
+  }
+
+  if (G_UNLIKELY (resample->need_discont)) {
+    GST_DEBUG ("marking this buffer with the DISCONT flag");
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+    resample->need_discont = FALSE;
+  }
+
+  return gst_speex_resample_process (resample, inbuf, outbuf);
+}
+
+static void
+gst_speex_resample_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstSpeexResample *resample;
+
+  resample = GST_SPEEX_RESAMPLE (object);
+
+  switch (prop_id) {
+    case PROP_QUALITY:
+      resample->quality = g_value_get_int (value);
+      GST_DEBUG ("new quality %d", resample->quality);
+
+      gst_speex_resample_update_state (resample, resample->channels,
+          resample->inrate, resample->outrate, resample->quality, resample->fp);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_speex_resample_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstSpeexResample *resample;
+
+  resample = GST_SPEEX_RESAMPLE (object);
+
+  switch (prop_id) {
+    case PROP_QUALITY:
+      g_value_set_int (value, resample->quality);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "speexresample", GST_RANK_NONE,
+          GST_TYPE_SPEEX_RESAMPLE)) {
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "speexresample",
+    "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
+    GST_PACKAGE_ORIGIN);
diff --git a/gst/speexresample/gstspeexresample.h b/gst/speexresample/gstspeexresample.h
new file mode 100644
index 00000000..68731289
--- /dev/null
+++ b/gst/speexresample/gstspeexresample.h
@@ -0,0 +1,80 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2007> Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+
+#ifndef __SPEEX_RESAMPLE_H__
+#define __SPEEX_RESAMPLE_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+
+#include "speex_resampler_wrapper.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_SPEEX_RESAMPLE \
+  (gst_speex_resample_get_type())
+#define GST_SPEEX_RESAMPLE(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResample))
+#define GST_SPEEX_RESAMPLE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResampleClass))
+#define GST_IS_SPEEX_RESAMPLE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_RESAMPLE))
+#define GST_IS_SPEEX_RESAMPLE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_RESAMPLE))
+
+typedef struct _GstSpeexResample GstSpeexResample;
+typedef struct _GstSpeexResampleClass GstSpeexResampleClass;
+
+/**
+ * GstSpeexResample:
+ *
+ * Opaque data structure.
+ */
+struct _GstSpeexResample {
+  GstBaseTransform element;
+
+  GstCaps *srccaps, *sinkcaps;
+
+  gboolean need_discont;
+
+  guint64 offset;
+  guint64 ts_offset;
+  GstClockTime next_ts;
+  GstClockTime prev_ts, prev_duration;
+  
+  gboolean fp;
+  int channels;
+  int inrate;
+  int outrate;
+  int quality;
+
+  SpeexResamplerState *state;
+};
+
+struct _GstSpeexResampleClass {
+  GstBaseTransformClass parent_class;
+};
+
+GType gst_speex_resample_get_type(void);
+
+G_END_DECLS
+
+#endif /* __SPEEX_RESAMPLE_H__ */
diff --git a/gst/speexresample/resample.c b/gst/speexresample/resample.c
new file mode 100644
index 00000000..f3c97fdd
--- /dev/null
+++ b/gst/speexresample/resample.c
@@ -0,0 +1,1310 @@
+/* Copyright (C) 2007 Jean-Marc Valin
+      
+   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.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#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 <math.h>
+
+#ifndef M_PI
+#define M_PI 3.14159263
+#endif
+
+#ifdef FIXED_POINT
+#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
+#else
+#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
+#endif
+
+/*#define float double*/
+#define FILTER_SIZE 64
+#define OVERSAMPLE 8
+
+#define IMAX(a,b) ((a) > (b) ? (a) : (b))
+#define IMIN(a,b) ((a) < (b) ? (a) : (b))
+
+#ifndef NULL
+#define NULL 0
+#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;
+  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 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 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 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 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 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
+{
+  double *table;
+  int oversample;
+};
+
+static struct FuncDef _KAISER12 = { kaiser12_table, 64 };
+
+#define KAISER12 (&_KAISER12)
+/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
+#define KAISER12 (&_KAISER12)*/
+static struct FuncDef _KAISER10 = { kaiser10_table, 32 };
+
+#define KAISER10 (&_KAISER10)
+static struct FuncDef _KAISER8 = { kaiser8_table, 32 };
+
+#define KAISER8 (&_KAISER8)
+static struct FuncDef _KAISER6 = { kaiser6_table, 32 };
+
+#define KAISER6 (&_KAISER6)
+
+struct QualityMapping
+{
+  int base_length;
+  int oversample;
+  float downsample_bandwidth;
+  float upsample_bandwidth;
+  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, 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, 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, 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)
+{
+  int N = st->filt_len;
+  int out_sample = 0;
+  spx_word16_t *mem;
+  int last_sample = st->last_sample[channel_index];
+  spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+
+  mem = st->mem + channel_index * st->mem_alloc_size;
+  while (!(last_sample >= (spx_int32_t) * in_len
+          || out_sample >= (spx_int32_t) * out_len)) {
+    int j;
+    spx_word32_t sum = 0;
+
+    /* We already have all the filter coefficients pre-computed in the table */
+    const spx_word16_t *ptr;
+
+    /* Do the memory part */
+    for (j = 0; last_sample - N + 1 + j < 0; j++) {
+      sum +=
+          MULT16_16 (mem[last_sample + j],
+          st->sinc_table[samp_frac_num * st->filt_len + j]);
+    }
+
+    /* Do the new part */
+    ptr = in + st->in_stride * (last_sample - N + 1 + j);
+    for (; j < N; j++) {
+      sum += MULT16_16 (*ptr, st->sinc_table[samp_frac_num * st->filt_len + j]);
+      ptr += st->in_stride;
+    }
+
+    *out = PSHR32 (sum, 15);
+    out += st->out_stride;
+    out_sample++;
+    last_sample += st->int_advance;
+    samp_frac_num += st->frac_advance;
+    if (samp_frac_num >= st->den_rate) {
+      samp_frac_num -= st->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)
+{
+  int N = st->filt_len;
+  int out_sample = 0;
+  spx_word16_t *mem;
+  int last_sample = st->last_sample[channel_index];
+  spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+
+  mem = st->mem + channel_index * st->mem_alloc_size;
+  while (!(last_sample >= (spx_int32_t) * in_len
+          || out_sample >= (spx_int32_t) * out_len)) {
+    int j;
+    double sum = 0;
+
+    /* We already have all the filter coefficients pre-computed in the table */
+    const spx_word16_t *ptr;
+
+    /* Do the memory part */
+    for (j = 0; last_sample - N + 1 + j < 0; j++) {
+      sum +=
+          MULT16_16 (mem[last_sample + j],
+          (double) st->sinc_table[samp_frac_num * st->filt_len + j]);
+    }
+
+    /* Do the new part */
+    ptr = in + st->in_stride * (last_sample - N + 1 + j);
+    for (; j < N; j++) {
+      sum +=
+          MULT16_16 (*ptr,
+          (double) st->sinc_table[samp_frac_num * st->filt_len + j]);
+      ptr += st->in_stride;
+    }
+
+    *out = sum;
+    out += st->out_stride;
+    out_sample++;
+    last_sample += st->int_advance;
+    samp_frac_num += st->frac_advance;
+    if (samp_frac_num >= st->den_rate) {
+      samp_frac_num -= st->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)
+{
+  int N = st->filt_len;
+  int out_sample = 0;
+  spx_word16_t *mem;
+  int last_sample = st->last_sample[channel_index];
+  spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+
+  mem = st->mem + channel_index * st->mem_alloc_size;
+  while (!(last_sample >= (spx_int32_t) * in_len
+          || out_sample >= (spx_int32_t) * out_len)) {
+    int j;
+    spx_word32_t sum = 0;
+
+    /* We need to interpolate the sinc filter */
+    spx_word32_t accum[4] = { 0.f, 0.f, 0.f, 0.f };
+    spx_word16_t interp[4];
+    const spx_word16_t *ptr;
+    int offset;
+    spx_word16_t frac;
+
+    offset = samp_frac_num * st->oversample / st->den_rate;
+#ifdef FIXED_POINT
+    frac =
+        PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
+        st->den_rate);
+#else
+    frac =
+        ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
+        st->den_rate;
+#endif
+    /* This code is written like this to make it easy to optimise with SIMD.
+       For most DSPs, it would be best to split the loops in two because most DSPs 
+       have only two accumulators */
+    for (j = 0; last_sample - N + 1 + j < 0; j++) {
+      spx_word16_t curr_mem = mem[last_sample + j];
+
+      accum[0] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
+      accum[1] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
+      accum[2] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset]);
+      accum[3] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
+    }
+    ptr = in + st->in_stride * (last_sample - N + 1 + j);
+    /* Do the new part */
+    for (; j < N; j++) {
+      spx_word16_t curr_in = *ptr;
+
+      ptr += st->in_stride;
+      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]);
+
+    *out = PSHR32 (sum, 15);
+    out += st->out_stride;
+    out_sample++;
+    last_sample += st->int_advance;
+    samp_frac_num += st->frac_advance;
+    if (samp_frac_num >= st->den_rate) {
+      samp_frac_num -= st->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)
+{
+  int N = st->filt_len;
+  int out_sample = 0;
+  spx_word16_t *mem;
+  int last_sample = st->last_sample[channel_index];
+  spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+
+  mem = st->mem + channel_index * st->mem_alloc_size;
+  while (!(last_sample >= (spx_int32_t) * in_len
+          || out_sample >= (spx_int32_t) * out_len)) {
+    int j;
+    spx_word32_t sum = 0;
+
+    /* We need to interpolate the sinc filter */
+    double accum[4] = { 0.f, 0.f, 0.f, 0.f };
+    float interp[4];
+    const spx_word16_t *ptr;
+    float alpha = ((float) samp_frac_num) / st->den_rate;
+    int offset = samp_frac_num * st->oversample / st->den_rate;
+    float frac = alpha * st->oversample - offset;
+
+    /* This code is written like this to make it easy to optimise with SIMD.
+       For most DSPs, it would be best to split the loops in two because most DSPs 
+       have only two accumulators */
+    for (j = 0; last_sample - N + 1 + j < 0; j++) {
+      double curr_mem = mem[last_sample + j];
+
+      accum[0] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
+      accum[1] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
+      accum[2] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset]);
+      accum[3] +=
+          MULT16_16 (curr_mem,
+          st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
+    }
+    ptr = in + st->in_stride * (last_sample - N + 1 + j);
+    /* Do the new part */
+    for (; j < N; j++) {
+      double curr_in = *ptr;
+
+      ptr += st->in_stride;
+      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 =
+        interp[0] * accum[0] + interp[1] * accum[1] + interp[2] * accum[2] +
+        interp[3] * accum[3];
+
+    *out = PSHR32 (sum, 15);
+    out += st->out_stride;
+    out_sample++;
+    last_sample += st->int_advance;
+    samp_frac_num += st->frac_advance;
+    if (samp_frac_num >= st->den_rate) {
+      samp_frac_num -= st->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 void
+update_filter (SpeexResamplerState * st)
+{
+  spx_uint32_t old_length;
+
+  old_length = st->filt_len;
+  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 down to make sure we have a multiple of 4 */
+    st->filt_len &= (~0x3);
+    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 */
+  if (st->den_rate <= st->oversample) {
+    spx_uint32_t i;
+
+    if (!st->sinc_table)
+      st->sinc_table =
+          (spx_word16_t *) speex_alloc (st->filt_len * st->den_rate *
+          sizeof (spx_word16_t));
+    else if (st->sinc_table_length < st->filt_len * st->den_rate) {
+      st->sinc_table =
+          (spx_word16_t *) speex_realloc (st->sinc_table,
+          st->filt_len * st->den_rate * sizeof (spx_word16_t));
+      st->sinc_table_length = st->filt_len * st->den_rate;
+    }
+    for (i = 0; i < st->den_rate; i++) {
+      spx_int32_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;
+
+    if (!st->sinc_table)
+      st->sinc_table =
+          (spx_word16_t *) speex_alloc ((st->filt_len * st->oversample +
+              8) * sizeof (spx_word16_t));
+    else if (st->sinc_table_length < st->filt_len * st->oversample + 8) {
+      st->sinc_table =
+          (spx_word16_t *) speex_realloc (st->sinc_table,
+          (st->filt_len * st->oversample + 8) * sizeof (spx_word16_t));
+      st->sinc_table_length = st->filt_len * st->oversample + 8;
+    }
+    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); */
+  }
+  st->int_advance = st->num_rate / st->den_rate;
+  st->frac_advance = st->num_rate % st->den_rate;
+
+
+  /* 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. */
+  if (!st->mem) {
+    spx_uint32_t i;
+
+    st->mem =
+        (spx_word16_t *) speex_alloc (st->nb_channels * (st->filt_len -
+            1) * sizeof (spx_word16_t));
+    for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
+      st->mem[i] = 0;
+    st->mem_alloc_size = st->filt_len - 1;
+    /*speex_warning("init filter"); */
+  } else if (!st->started) {
+    spx_uint32_t i;
+
+    st->mem =
+        (spx_word16_t *) speex_realloc (st->mem,
+        st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t));
+    for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
+      st->mem[i] = 0;
+    st->mem_alloc_size = st->filt_len - 1;
+    /*speex_warning("reinit filter"); */
+  } else if (st->filt_len > old_length) {
+    spx_int32_t i;
+
+    /* Increase the filter length */
+    /*speex_warning("increase filter size"); */
+    int old_alloc_size = st->mem_alloc_size;
+
+    if (st->filt_len - 1 > st->mem_alloc_size) {
+      st->mem =
+          (spx_word16_t *) speex_realloc (st->mem,
+          st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t));
+      st->mem_alloc_size = st->filt_len - 1;
+    }
+    for (i = st->nb_channels - 1; i >= 0; i--) {
+      spx_int32_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 - 2 + st->magic_samples[i]; j >= 0; 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;
+    }
+  }
+
+}
+
+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);
+}
+
+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;
+
+  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;
+
+  /* Per channel data */
+  st->last_sample = (spx_int32_t *) speex_alloc (nb_channels * sizeof (int));
+  st->magic_samples = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
+  st->samp_frac_num = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
+  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);
+
+
+  update_filter (st);
+
+  st->initialised = 1;
+  if (err)
+    *err = RESAMPLER_ERR_SUCCESS;
+
+  return st;
+}
+
+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, const spx_word16_t * in, spx_uint32_t * in_len,
+    spx_word16_t * out, spx_uint32_t * out_len)
+{
+  int j = 0;
+  int N = st->filt_len;
+  int out_sample = 0;
+  spx_word16_t *mem;
+  spx_uint32_t tmp_out_len = 0;
+
+  mem = st->mem + channel_index * st->mem_alloc_size;
+  st->started = 1;
+
+  /* Handle the case where we have samples left from a reduction in filter length */
+  if (st->magic_samples[channel_index]) {
+    int istride_save;
+    spx_uint32_t tmp_in_len;
+    spx_uint32_t tmp_magic;
+
+    istride_save = st->in_stride;
+    tmp_in_len = st->magic_samples[channel_index];
+    tmp_out_len = *out_len;
+    /* magic_samples needs to be set to zero to avoid infinite recursion */
+    tmp_magic = st->magic_samples[channel_index];
+    st->magic_samples[channel_index] = 0;
+    st->in_stride = 1;
+    speex_resampler_process_native (st, channel_index, mem + N - 1, &tmp_in_len,
+        out, &tmp_out_len);
+    st->in_stride = istride_save;
+    /*speex_warning_int("extra samples:", tmp_out_len); */
+    /* If we couldn't process all "magic" input samples, save the rest for next time */
+    if (tmp_in_len < tmp_magic) {
+      spx_uint32_t i;
+
+      st->magic_samples[channel_index] = tmp_magic - tmp_in_len;
+      for (i = 0; i < st->magic_samples[channel_index]; i++)
+        mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
+    }
+    out += tmp_out_len * st->out_stride;
+    *out_len -= tmp_out_len;
+  }
+
+  /* Call the right resampler through the function ptr */
+  out_sample = st->resampler_ptr (st, channel_index, in, 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 + tmp_out_len;
+  st->last_sample[channel_index] -= *in_len;
+
+  for (j = 0; j < N - 1 - (spx_int32_t) * in_len; j++)
+    mem[j] = mem[j + *in_len];
+  for (; j < N - 1; j++)
+    mem[j] = in[st->in_stride * (j + *in_len - N + 1)];
+
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+#define FIXED_STACK_ALLOC 1024
+
+#ifdef FIXED_POINT
+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)
+{
+  spx_uint32_t i;
+  int istride_save, ostride_save;
+
+#ifdef VAR_ARRAYS
+  spx_word16_t x[*in_len];
+  spx_word16_t y[*out_len];
+
+  /*VARDECL(spx_word16_t *x);
+     VARDECL(spx_word16_t *y);
+     ALLOC(x, *in_len, spx_word16_t);
+     ALLOC(y, *out_len, spx_word16_t); */
+  istride_save = st->in_stride;
+  ostride_save = st->out_stride;
+  for (i = 0; i < *in_len; i++)
+    x[i] = WORD2INT (in[i * st->in_stride]);
+  st->in_stride = st->out_stride = 1;
+  speex_resampler_process_native (st, channel_index, x, in_len, y, out_len);
+  st->in_stride = istride_save;
+  st->out_stride = ostride_save;
+  for (i = 0; i < *out_len; i++)
+    out[i * st->out_stride] = y[i];
+#else
+  spx_word16_t x[FIXED_STACK_ALLOC];
+  spx_word16_t y[FIXED_STACK_ALLOC];
+  spx_uint32_t ilen = *in_len, olen = *out_len;
+
+  istride_save = st->in_stride;
+  ostride_save = st->out_stride;
+  while (ilen && olen) {
+    spx_uint32_t ichunk, ochunk;
+
+    ichunk = ilen;
+    ochunk = olen;
+    if (ichunk > FIXED_STACK_ALLOC)
+      ichunk = FIXED_STACK_ALLOC;
+    if (ochunk > FIXED_STACK_ALLOC)
+      ochunk = FIXED_STACK_ALLOC;
+    for (i = 0; i < ichunk; i++)
+      x[i] = WORD2INT (in[i * st->in_stride]);
+    st->in_stride = st->out_stride = 1;
+    speex_resampler_process_native (st, channel_index, x, &ichunk, y, &ochunk);
+    st->in_stride = istride_save;
+    st->out_stride = ostride_save;
+    for (i = 0; i < ochunk; i++)
+      out[i * st->out_stride] = y[i];
+    out += ochunk;
+    in += ichunk;
+    ilen -= ichunk;
+    olen -= ochunk;
+  }
+  *in_len -= ilen;
+  *out_len -= olen;
+#endif
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+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)
+{
+  return speex_resampler_process_native (st, channel_index, in, in_len, out,
+      out_len);
+}
+#else
+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)
+{
+  return speex_resampler_process_native (st, channel_index, in, in_len, out,
+      out_len);
+}
+
+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)
+{
+  spx_uint32_t i;
+  int istride_save, ostride_save;
+
+#ifdef VAR_ARRAYS
+  spx_word16_t x[*in_len];
+  spx_word16_t y[*out_len];
+
+  /*VARDECL(spx_word16_t *x);
+     VARDECL(spx_word16_t *y);
+     ALLOC(x, *in_len, spx_word16_t);
+     ALLOC(y, *out_len, spx_word16_t); */
+  istride_save = st->in_stride;
+  ostride_save = st->out_stride;
+  for (i = 0; i < *in_len; i++)
+    x[i] = in[i * st->in_stride];
+  st->in_stride = st->out_stride = 1;
+  speex_resampler_process_native (st, channel_index, x, in_len, y, out_len);
+  st->in_stride = istride_save;
+  st->out_stride = ostride_save;
+  for (i = 0; i < *out_len; i++)
+    out[i * st->out_stride] = WORD2INT (y[i]);
+#else
+  spx_word16_t x[FIXED_STACK_ALLOC];
+  spx_word16_t y[FIXED_STACK_ALLOC];
+  spx_uint32_t ilen = *in_len, olen = *out_len;
+
+  istride_save = st->in_stride;
+  ostride_save = st->out_stride;
+  while (ilen && olen) {
+    spx_uint32_t ichunk, ochunk;
+
+    ichunk = ilen;
+    ochunk = olen;
+    if (ichunk > FIXED_STACK_ALLOC)
+      ichunk = FIXED_STACK_ALLOC;
+    if (ochunk > FIXED_STACK_ALLOC)
+      ochunk = FIXED_STACK_ALLOC;
+    for (i = 0; i < ichunk; i++)
+      x[i] = in[i * st->in_stride];
+    st->in_stride = st->out_stride = 1;
+    speex_resampler_process_native (st, channel_index, x, &ichunk, y, &ochunk);
+    st->in_stride = istride_save;
+    st->out_stride = ostride_save;
+    for (i = 0; i < ochunk; i++)
+      out[i * st->out_stride] = WORD2INT (y[i]);
+    out += ochunk;
+    in += ichunk;
+    ilen -= ichunk;
+    olen -= ochunk;
+  }
+  *in_len -= ilen;
+  *out_len -= olen;
+#endif
+  return RESAMPLER_ERR_SUCCESS;
+}
+#endif
+
+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_len = *out_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_len;
+    speex_resampler_process_float (st, i, in + i, in_len, out + i, out_len);
+  }
+  st->in_stride = istride_save;
+  st->out_stride = ostride_save;
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+
+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_len = *out_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_len;
+    speex_resampler_process_int (st, i, in + i, in_len, out + i, out_len);
+  }
+  st->in_stride = istride_save;
+  st->out_stride = ostride_save;
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+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);
+}
+
+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;
+}
+
+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)
+    update_filter (st);
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+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;
+}
+
+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)
+    update_filter (st);
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+void
+speex_resampler_get_quality (SpeexResamplerState * st, int *quality)
+{
+  *quality = st->quality;
+}
+
+void
+speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride)
+{
+  st->in_stride = stride;
+}
+
+void
+speex_resampler_get_input_stride (SpeexResamplerState * st,
+    spx_uint32_t * stride)
+{
+  *stride = st->in_stride;
+}
+
+void
+speex_resampler_set_output_stride (SpeexResamplerState * st,
+    spx_uint32_t stride)
+{
+  st->out_stride = stride;
+}
+
+void
+speex_resampler_get_output_stride (SpeexResamplerState * st,
+    spx_uint32_t * stride)
+{
+  *stride = st->out_stride;
+}
+
+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;
+}
+
+int
+speex_resampler_reset_mem (SpeexResamplerState * st)
+{
+  spx_uint32_t i;
+
+  for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
+    st->mem[i] = 0;
+  return RESAMPLER_ERR_SUCCESS;
+}
+
+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/gst/speexresample/speex_resampler.h b/gst/speexresample/speex_resampler.h
new file mode 100644
index 00000000..1dde54ac
--- /dev/null
+++ b/gst/speexresample/speex_resampler.h
@@ -0,0 +1,325 @@
+/* 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
+
+#ifdef OUTSIDE_SPEEX
+
+#include <glib.h>
+
+/********* 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 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_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 gint16
+#define spx_int32_t gint32
+#define spx_uint16_t guint16
+#define spx_uint32_t guint32
+      
+#else /* OUTSIDE_SPEEX */
+
+#include "speex/speex_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);
+
+/** 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/gst/speexresample/speex_resampler_float.c b/gst/speexresample/speex_resampler_float.c
new file mode 100644
index 00000000..281e52d3
--- /dev/null
+++ b/gst/speexresample/speex_resampler_float.c
@@ -0,0 +1,24 @@
+/* GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#define FLOATING_POINT
+#define OUTSIDE_SPEEX
+#define RANDOM_PREFIX resample_float
+
+#include "resample.c"
diff --git a/gst/speexresample/speex_resampler_int.c b/gst/speexresample/speex_resampler_int.c
new file mode 100644
index 00000000..c992f0a6
--- /dev/null
+++ b/gst/speexresample/speex_resampler_int.c
@@ -0,0 +1,24 @@
+/* GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#define FIXED_POINT 1
+#define OUTSIDE_SPEEX 1
+#define RANDOM_PREFIX resample_int
+
+#include "resample.c"
diff --git a/gst/speexresample/speex_resampler_wrapper.h b/gst/speexresample/speex_resampler_wrapper.h
new file mode 100644
index 00000000..25f5576d
--- /dev/null
+++ b/gst/speexresample/speex_resampler_wrapper.h
@@ -0,0 +1,80 @@
+/* GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __SPEEX_RESAMPLER_WRAPPER_H__
+#define __SPEEX_RESAMPLER_WRAPPER_H__
+
+#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
+};
+
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+SpeexResamplerState *resample_float_resampler_init (guint32 nb_channels,
+    guint32 in_rate, guint32 out_rate, gint quality, gint * err);
+SpeexResamplerState *resample_int_resampler_init (guint32 nb_channels,
+    guint32 in_rate, guint32 out_rate, gint quality, gint * err);
+
+#define resample_resampler_destroy resample_int_resampler_destroy
+void resample_resampler_destroy (SpeexResamplerState * st);
+
+int resample_float_resampler_process_interleaved_float (SpeexResamplerState *
+    st, const gfloat * in, guint32 * in_len, gfloat * out, guint32 * out_len);
+int resample_int_resampler_process_interleaved_int (SpeexResamplerState * st,
+    const gint16 * in, guint32 * in_len, gint16 * out, guint32 * out_len);
+
+int resample_float_resampler_set_rate (SpeexResamplerState * st,
+    guint32 in_rate, guint32 out_rate);
+int resample_int_resampler_set_rate (SpeexResamplerState * st,
+    guint32 in_rate, guint32 out_rate);
+
+void resample_float_resampler_get_rate (SpeexResamplerState * st,
+    guint32 * in_rate, guint32 * out_rate);
+void resample_int_resampler_get_rate (SpeexResamplerState * st,
+    guint32 * in_rate, guint32 * out_rate);
+
+void resample_float_resampler_get_ratio (SpeexResamplerState * st,
+    guint32 * ratio_num, guint32 * ratio_den);
+void resample_int_resampler_get_ratio (SpeexResamplerState * st,
+    guint32 * ratio_num, guint32 * ratio_den);
+
+int resample_float_resampler_set_quality (SpeexResamplerState * st,
+    gint quality);
+int resample_int_resampler_set_quality (SpeexResamplerState * st, gint quality);
+
+int resample_float_resampler_reset_mem (SpeexResamplerState * st);
+int resample_int_resampler_reset_mem (SpeexResamplerState * st);
+
+#define resample_resampler_strerror resample_int_resampler_strerror
+const char *resample_resampler_strerror (gint err);
+
+#endif /* __SPEEX_RESAMPLER_WRAPPER_H__ */