--- /dev/null
+/* 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
--- /dev/null
+/* 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
--- /dev/null
+/* 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);
--- /dev/null
+/* 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__ */
--- /dev/null
+/* 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.";
+ }
+}
--- /dev/null
+/* 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
--- /dev/null
+/* 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"
--- /dev/null
+/* 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"
--- /dev/null
+/* 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__ */
projects / platform / upstream / gst-plugins-base.git / commitdiff