diff options
Diffstat (limited to 'gst')
| -rw-r--r-- | gst/speexresample/arch.h | 243 | ||||
| -rw-r--r-- | gst/speexresample/fixed_generic.h | 106 | ||||
| -rw-r--r-- | gst/speexresample/gstspeexresample.c | 733 | ||||
| -rw-r--r-- | gst/speexresample/gstspeexresample.h | 80 | ||||
| -rw-r--r-- | gst/speexresample/resample.c | 1310 | ||||
| -rw-r--r-- | gst/speexresample/speex_resampler.h | 325 | ||||
| -rw-r--r-- | gst/speexresample/speex_resampler_float.c | 24 | ||||
| -rw-r--r-- | gst/speexresample/speex_resampler_int.c | 24 | ||||
| -rw-r--r-- | gst/speexresample/speex_resampler_wrapper.h | 80 |
9 files changed, 2925 insertions, 0 deletions
diff --git a/gst/speexresample/arch.h b/gst/speexresample/arch.h new file mode 100644 index 00000000..f213e683 --- /dev/null +++ b/gst/speexresample/arch.h @@ -0,0 +1,243 @@ +/* Copyright (C) 2003 Jean-Marc Valin */ +/** + @file arch.h + @brief Various architecture definitions Speex +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef ARCH_H +#define ARCH_H + +#ifndef SPEEX_VERSION +#define SPEEX_MAJOR_VERSION 1 /**< Major Speex version. */ +#define SPEEX_MINOR_VERSION 1 /**< Minor Speex version. */ +#define SPEEX_MICRO_VERSION 15 /**< Micro Speex version. */ +#define SPEEX_EXTRA_VERSION "" /**< Extra Speex version. */ +#define SPEEX_VERSION "speex-1.2beta3" /**< Speex version string. */ +#endif + +/* A couple test to catch stupid option combinations */ +#ifdef FIXED_POINT + +#ifdef FLOATING_POINT +#error You cannot compile as floating point and fixed point at the same time +#endif +#ifdef _USE_SSE +#error SSE is only for floating-point +#endif +#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM)) +#error Make up your mind. What CPU do you have? +#endif +#ifdef VORBIS_PSYCHO +#error Vorbis-psy model currently not implemented in fixed-point +#endif + +#else + +#ifndef FLOATING_POINT +#error You now need to define either FIXED_POINT or FLOATING_POINT +#endif +#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM) +#error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions? +#endif +#ifdef FIXED_POINT_DEBUG +#error "Don't you think enabling fixed-point is a good thing to do if you want to debug that?" +#endif + + +#endif + +#ifndef OUTSIDE_SPEEX +#include "speex/speex_types.h" +#endif + +#ifndef ABS +#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ +#endif +#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 16-bit value. */ +#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 16-bit value. */ +#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */ +#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 32-bit value. */ +#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 32-bit value. */ +#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */ + +#ifdef FIXED_POINT + +typedef spx_int16_t spx_word16_t; +typedef spx_int32_t spx_word32_t; +typedef spx_word32_t spx_mem_t; +typedef spx_word16_t spx_coef_t; +typedef spx_word16_t spx_lsp_t; +typedef spx_word32_t spx_sig_t; + +#define Q15ONE 32767 + +#define LPC_SCALING 8192 +#define SIG_SCALING 16384 +#define LSP_SCALING 8192. +#define GAMMA_SCALING 32768. +#define GAIN_SCALING 64 +#define GAIN_SCALING_1 0.015625 + +#define LPC_SHIFT 13 +#define LSP_SHIFT 13 +#define SIG_SHIFT 14 +#define GAIN_SHIFT 6 + +#define VERY_SMALL 0 +#define VERY_LARGE32 ((spx_word32_t)2147483647) +#define VERY_LARGE16 ((spx_word16_t)32767) +#define Q15_ONE ((spx_word16_t)32767) + + +#ifdef FIXED_DEBUG +#include "fixed_debug.h" +#else + +#include "fixed_generic.h" + +#ifdef ARM5E_ASM +#include "fixed_arm5e.h" +#elif defined (ARM4_ASM) +#include "fixed_arm4.h" +#elif defined (ARM5E_ASM) +#include "fixed_arm5e.h" +#elif defined (BFIN_ASM) +#include "fixed_bfin.h" +#endif + +#endif + + +#else + +typedef float spx_mem_t; +typedef float spx_coef_t; +typedef float spx_lsp_t; +typedef float spx_sig_t; +typedef float spx_word16_t; +typedef float spx_word32_t; + +#define Q15ONE 1.0f +#define LPC_SCALING 1.f +#define SIG_SCALING 1.f +#define LSP_SCALING 1.f +#define GAMMA_SCALING 1.f +#define GAIN_SCALING 1.f +#define GAIN_SCALING_1 1.f + + +#define VERY_SMALL 1e-15f +#define VERY_LARGE32 1e15f +#define VERY_LARGE16 1e15f +#define Q15_ONE ((spx_word16_t)1.f) + +#define QCONST16(x,bits) (x) +#define QCONST32(x,bits) (x) + +#define NEG16(x) (-(x)) +#define NEG32(x) (-(x)) +#define EXTRACT16(x) (x) +#define EXTEND32(x) (x) +#define SHR16(a,shift) (a) +#define SHL16(a,shift) (a) +#define SHR32(a,shift) (a) +#define SHL32(a,shift) (a) +#define PSHR16(a,shift) (a) +#define PSHR32(a,shift) (a) +#define VSHR32(a,shift) (a) +#define SATURATE16(x,a) (x) +#define SATURATE32(x,a) (x) + +#define PSHR(a,shift) (a) +#define SHR(a,shift) (a) +#define SHL(a,shift) (a) +#define SATURATE(x,a) (x) + +#define ADD16(a,b) ((a)+(b)) +#define SUB16(a,b) ((a)-(b)) +#define ADD32(a,b) ((a)+(b)) +#define SUB32(a,b) ((a)-(b)) +#define MULT16_16_16(a,b) ((a)*(b)) +#define MULT16_16(a,b) ((spx_word32_t)(a)*(spx_word32_t)(b)) +#define MAC16_16(c,a,b) ((c)+(spx_word32_t)(a)*(spx_word32_t)(b)) + +#define MULT16_32_Q11(a,b) ((a)*(b)) +#define MULT16_32_Q13(a,b) ((a)*(b)) +#define MULT16_32_Q14(a,b) ((a)*(b)) +#define MULT16_32_Q15(a,b) ((a)*(b)) +#define MULT16_32_P15(a,b) ((a)*(b)) + +#define MAC16_32_Q11(c,a,b) ((c)+(a)*(b)) +#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b)) + +#define MAC16_16_Q11(c,a,b) ((c)+(a)*(b)) +#define MAC16_16_Q13(c,a,b) ((c)+(a)*(b)) +#define MAC16_16_P13(c,a,b) ((c)+(a)*(b)) +#define MULT16_16_Q11_32(a,b) ((a)*(b)) +#define MULT16_16_Q13(a,b) ((a)*(b)) +#define MULT16_16_Q14(a,b) ((a)*(b)) +#define MULT16_16_Q15(a,b) ((a)*(b)) +#define MULT16_16_P15(a,b) ((a)*(b)) +#define MULT16_16_P13(a,b) ((a)*(b)) +#define MULT16_16_P14(a,b) ((a)*(b)) + +#define DIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b)) +#define PDIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b)) +#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b)) +#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b)) + + +#endif + + +#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + +/* 2 on TI C5x DSP */ +#define BYTES_PER_CHAR 2 +#define BITS_PER_CHAR 16 +#define LOG2_BITS_PER_CHAR 4 + +#else + +#define BYTES_PER_CHAR 1 +#define BITS_PER_CHAR 8 +#define LOG2_BITS_PER_CHAR 3 + +#endif + + + +#ifdef FIXED_DEBUG +long long spx_mips=0; +#endif + + +#endif diff --git a/gst/speexresample/fixed_generic.h b/gst/speexresample/fixed_generic.h new file mode 100644 index 00000000..2948177c --- /dev/null +++ b/gst/speexresample/fixed_generic.h @@ -0,0 +1,106 @@ +/* Copyright (C) 2003 Jean-Marc Valin */ +/** + @file fixed_generic.h + @brief Generic fixed-point operations +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef FIXED_GENERIC_H +#define FIXED_GENERIC_H + +#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits)))) +#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits)))) + +#define NEG16(x) (-(x)) +#define NEG32(x) (-(x)) +#define EXTRACT16(x) ((spx_word16_t)(x)) +#define EXTEND32(x) ((spx_word32_t)(x)) +#define SHR16(a,shift) ((a) >> (shift)) +#define SHL16(a,shift) ((a) << (shift)) +#define SHR32(a,shift) ((a) >> (shift)) +#define SHL32(a,shift) ((a) << (shift)) +#define PSHR16(a,shift) (SHR16((a)+((1<<((shift))>>1)),shift)) +#define PSHR32(a,shift) (SHR32((a)+((1<<((shift))>>1)),shift)) +#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) +#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) +#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) + +#define SHR(a,shift) ((a) >> (shift)) +#define SHL(a,shift) ((spx_word32_t)(a) << (shift)) +#define PSHR(a,shift) (SHR((a)+((1<<((shift))>>1)),shift)) +#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) + + +#define ADD16(a,b) ((spx_word16_t)((spx_word16_t)(a)+(spx_word16_t)(b))) +#define SUB16(a,b) ((spx_word16_t)(a)-(spx_word16_t)(b)) +#define ADD32(a,b) ((spx_word32_t)(a)+(spx_word32_t)(b)) +#define SUB32(a,b) ((spx_word32_t)(a)-(spx_word32_t)(b)) + + +/* result fits in 16 bits */ +#define MULT16_16_16(a,b) ((((spx_word16_t)(a))*((spx_word16_t)(b)))) + +/* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */ +#define MULT16_16(a,b) (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b))) + +#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b)))) +#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12)) +#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13)) +#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14)) + +#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)) +#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))) + +#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15)) +#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)) +#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))) + + +#define MAC16_16_Q11(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),11))) +#define MAC16_16_Q13(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),13))) +#define MAC16_16_P13(c,a,b) (ADD32((c),SHR(ADD32(4096,MULT16_16((a),(b))),13))) + +#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11)) +#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13)) +#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14)) +#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15)) + +#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13)) +#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14)) +#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15)) + +#define MUL_16_32_R15(a,bh,bl) ADD32(MULT16_16((a),(bh)), SHR(MULT16_16((a),(bl)),15)) + +#define DIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a))/((spx_word16_t)(b)))) +#define PDIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word16_t)(b)))) +#define DIV32(a,b) (((spx_word32_t)(a))/((spx_word32_t)(b))) +#define PDIV32(a,b) (((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word32_t)(b))) + +#endif diff --git a/gst/speexresample/gstspeexresample.c b/gst/speexresample/gstspeexresample.c new file mode 100644 index 00000000..307b8180 --- /dev/null +++ b/gst/speexresample/gstspeexresample.c @@ -0,0 +1,733 @@ +/* GStreamer + * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu> + * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org> + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-speexresample + * + * <refsect2> + * speexresample resamples raw audio buffers to different sample rates using + * a configurable windowing function to enhance quality. + * <title>Example launch line</title> + * <para> + * <programlisting> + * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! speexresample ! audio/x-raw-int, rate=8000 ! alsasink + * </programlisting> + * Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa. + * To create the Ogg/Vorbis file refer to the documentation of vorbisenc. + * </para> + * </refsect2> + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <string.h> +#include <math.h> + +#include "gstspeexresample.h" +#include <gst/audio/audio.h> +#include <gst/base/gstbasetransform.h> + +GST_DEBUG_CATEGORY (speex_resample_debug); +#define GST_CAT_DEFAULT speex_resample_debug + +enum +{ + PROP_0, + PROP_QUALITY +}; + +#define SUPPORTED_CAPS \ +GST_STATIC_CAPS ( \ + "audio/x-raw-float, " \ + "rate = (int) [ 1, MAX ], " \ + "channels = (int) [ 1, MAX ], " \ + "endianness = (int) BYTE_ORDER, " \ + "width = (int) 32; " \ + "audio/x-raw-int, " \ + "rate = (int) [ 1, MAX ], " \ + "channels = (int) [ 1, MAX ], " \ + "endianness = (int) BYTE_ORDER, " \ + "width = (int) 16, " \ + "depth = (int) 16, " \ + "signed = (boolean) true" \ +) + +static GstStaticPadTemplate gst_speex_resample_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS); + +static GstStaticPadTemplate gst_speex_resample_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS); + +static void gst_speex_resample_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_speex_resample_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +/* vmethods */ +static gboolean gst_speex_resample_get_unit_size (GstBaseTransform * base, + GstCaps * caps, guint * size); +static GstCaps *gst_speex_resample_transform_caps (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps); +static gboolean gst_speex_resample_transform_size (GstBaseTransform * trans, + GstPadDirection direction, GstCaps * incaps, guint insize, + GstCaps * outcaps, guint * outsize); +static gboolean gst_speex_resample_set_caps (GstBaseTransform * base, + GstCaps * incaps, GstCaps * outcaps); +static GstFlowReturn gst_speex_resample_transform (GstBaseTransform * base, + GstBuffer * inbuf, GstBuffer * outbuf); +static gboolean gst_speex_resample_event (GstBaseTransform * base, + GstEvent * event); +static gboolean gst_speex_resample_start (GstBaseTransform * base); +static gboolean gst_speex_resample_stop (GstBaseTransform * base); + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (speex_resample_debug, "speex_resample", 0, "audio resampling element"); + +GST_BOILERPLATE_FULL (GstSpeexResample, gst_speex_resample, GstBaseTransform, + GST_TYPE_BASE_TRANSFORM, DEBUG_INIT); + +static void +gst_speex_resample_base_init (gpointer g_class) +{ + GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_speex_resample_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_speex_resample_sink_template)); + + gst_element_class_set_details_simple (gstelement_class, "Audio resampler", + "Filter/Converter/Audio", "Resamples audio", + "Sebastian Dröge <slomo@circular-chaos.org>"); +} + +static void +gst_speex_resample_class_init (GstSpeexResampleClass * klass) +{ + GObjectClass *gobject_class = (GObjectClass *) klass; + + gobject_class->set_property = gst_speex_resample_set_property; + gobject_class->get_property = gst_speex_resample_get_property; + + g_object_class_install_property (gobject_class, PROP_QUALITY, + g_param_spec_int ("quality", "Quality", "Resample quality with 0 being " + "the lowest and 10 being the best", + SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX, + SPEEX_RESAMPLER_QUALITY_DEFAULT, + G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); + + GST_BASE_TRANSFORM_CLASS (klass)->start = + GST_DEBUG_FUNCPTR (gst_speex_resample_start); + GST_BASE_TRANSFORM_CLASS (klass)->stop = + GST_DEBUG_FUNCPTR (gst_speex_resample_stop); + GST_BASE_TRANSFORM_CLASS (klass)->transform_size = + GST_DEBUG_FUNCPTR (gst_speex_resample_transform_size); + GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = + GST_DEBUG_FUNCPTR (gst_speex_resample_get_unit_size); + GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = + GST_DEBUG_FUNCPTR (gst_speex_resample_transform_caps); + GST_BASE_TRANSFORM_CLASS (klass)->set_caps = + GST_DEBUG_FUNCPTR (gst_speex_resample_set_caps); + GST_BASE_TRANSFORM_CLASS (klass)->transform = + GST_DEBUG_FUNCPTR (gst_speex_resample_transform); + GST_BASE_TRANSFORM_CLASS (klass)->event = + GST_DEBUG_FUNCPTR (gst_speex_resample_event); + + GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE; +} + +static void +gst_speex_resample_init (GstSpeexResample * resample, + GstSpeexResampleClass * klass) +{ + resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT; + + resample->need_discont = FALSE; +} + +/* vmethods */ +static gboolean +gst_speex_resample_start (GstBaseTransform * base) +{ + GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); + + resample->ts_offset = -1; + resample->offset = -1; + resample->next_ts = -1; + + return TRUE; +} + +static gboolean +gst_speex_resample_stop (GstBaseTransform * base) +{ + GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); + + if (resample->state) { + resample_resampler_destroy (resample->state); + resample->state = NULL; + } + + gst_caps_replace (&resample->sinkcaps, NULL); + gst_caps_replace (&resample->srccaps, NULL); + + return TRUE; +} + +static gboolean +gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps, + guint * size) +{ + gint width, channels; + GstStructure *structure; + gboolean ret; + + g_return_val_if_fail (size != NULL, FALSE); + + /* this works for both float and int */ + structure = gst_caps_get_structure (caps, 0); + ret = gst_structure_get_int (structure, "width", &width); + ret &= gst_structure_get_int (structure, "channels", &channels); + g_return_val_if_fail (ret, FALSE); + + *size = width * channels / 8; + + return TRUE; +} + +static GstCaps * +gst_speex_resample_transform_caps (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps) +{ + GstCaps *res; + GstStructure *structure; + + /* transform caps gives one single caps so we can just replace + * the rate property with our range. */ + res = gst_caps_copy (caps); + structure = gst_caps_get_structure (res, 0); + gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); + + return res; +} + +static SpeexResamplerState * +gst_speex_resample_init_state (guint channels, guint inrate, guint outrate, + guint quality, gboolean fp) +{ + SpeexResamplerState *ret = NULL; + gint err = RESAMPLER_ERR_SUCCESS; + + if (fp) + ret = + resample_float_resampler_init (channels, inrate, outrate, quality, + &err); + else + ret = + resample_int_resampler_init (channels, inrate, outrate, quality, &err); + + if (err != RESAMPLER_ERR_SUCCESS) { + GST_ERROR ("Failed to create resampler state: %s", + resample_resampler_strerror (err)); + return NULL; + } + + return ret; +} + +static gboolean +gst_speex_resample_update_state (GstSpeexResample * resample, gint channels, + gint inrate, gint outrate, gint quality, gboolean fp) +{ + gboolean ret = TRUE; + + if (resample->state == NULL) { + ret = TRUE; + } else if (resample->channels != channels || fp != resample->fp) { + resample_resampler_destroy (resample->state); + resample->state = + gst_speex_resample_init_state (channels, inrate, outrate, quality, fp); + + ret = (resample->state != NULL); + } else if (resample->inrate != inrate || resample->outrate != outrate) { + gint err = RESAMPLER_ERR_SUCCESS; + + if (fp) + err = + resample_float_resampler_set_rate (resample->state, inrate, outrate); + else + err = resample_int_resampler_set_rate (resample->state, inrate, outrate); + + if (err != RESAMPLER_ERR_SUCCESS) + GST_ERROR ("Failed to update rate: %s", + resample_resampler_strerror (err)); + + ret = (err == RESAMPLER_ERR_SUCCESS); + } else if (quality != resample->quality) { + gint err = RESAMPLER_ERR_SUCCESS; + + if (fp) + err = resample_float_resampler_set_quality (resample->state, quality); + else + err = resample_int_resampler_set_quality (resample->state, quality); + + if (err != RESAMPLER_ERR_SUCCESS) + GST_ERROR ("Failed to update quality: %s", + resample_resampler_strerror (err)); + + ret = (err == RESAMPLER_ERR_SUCCESS); + } + + resample->channels = channels; + resample->fp = fp; + resample->quality = quality; + resample->inrate = inrate; + resample->outrate = outrate; + + return ret; +} + +static void +gst_speex_resample_reset_state (GstSpeexResample * resample) +{ + if (resample->state && resample->fp) + resample_float_resampler_reset_mem (resample->state); + else if (resample->state && !resample->fp) + resample_int_resampler_reset_mem (resample->state); +} + +static gboolean +gst_speex_resample_parse_caps (GstCaps * incaps, + GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate, + gboolean * fp) +{ + GstStructure *structure; + gboolean ret; + gint myinrate, myoutrate, mychannels; + gboolean myfp; + + GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %" + GST_PTR_FORMAT, incaps, outcaps); + + structure = gst_caps_get_structure (incaps, 0); + + if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) + myfp = TRUE; + else + myfp = FALSE; + + ret = gst_structure_get_int (structure, "rate", &myinrate); + ret &= gst_structure_get_int (structure, "channels", &mychannels); + if (!ret) + goto no_in_rate_channels; + + structure = gst_caps_get_structure (outcaps, 0); + ret = gst_structure_get_int (structure, "rate", &myoutrate); + if (!ret) + goto no_out_rate; + + if (channels) + *channels = mychannels; + if (inrate) + *inrate = myinrate; + if (outrate) + *outrate = myoutrate; + + if (fp) + *fp = myfp; + + return TRUE; + + /* ERRORS */ +no_in_rate_channels: + { + GST_DEBUG ("could not get input rate and channels"); + return FALSE; + } +no_out_rate: + { + GST_DEBUG ("could not get output rate"); + return FALSE; + } +} + +static gboolean +gst_speex_resample_transform_size (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps, + guint * othersize) +{ + GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); + SpeexResamplerState *state; + GstCaps *srccaps, *sinkcaps; + gboolean use_internal = FALSE; /* whether we use the internal state */ + gboolean ret = TRUE; + guint32 ratio_den, ratio_num; + gboolean fp; + + GST_LOG ("asked to transform size %d in direction %s", + size, direction == GST_PAD_SINK ? "SINK" : "SRC"); + if (direction == GST_PAD_SINK) { + sinkcaps = caps; + srccaps = othercaps; + } else { + sinkcaps = othercaps; + srccaps = caps; + } + + /* if the caps are the ones that _set_caps got called with; we can use + * our own state; otherwise we'll have to create a state */ + if (resample->state && gst_caps_is_equal (sinkcaps, resample->sinkcaps) && + gst_caps_is_equal (srccaps, resample->srccaps)) { + use_internal = TRUE; + state = resample->state; + fp = resample->fp; + } else { + gint inrate, outrate, channels; + + GST_DEBUG ("Can't use internal state, creating state"); + + ret = + gst_speex_resample_parse_caps (caps, othercaps, &channels, &inrate, + &outrate, &fp); + + if (!ret) { + GST_ERROR ("Wrong caps"); + return FALSE; + } + + state = gst_speex_resample_init_state (channels, inrate, outrate, 0, TRUE); + if (!state) + return FALSE; + } + + if (resample->fp || use_internal) + resample_float_resampler_get_ratio (state, &ratio_num, &ratio_den); + else + resample_int_resampler_get_ratio (state, &ratio_num, &ratio_den); + + if (direction == GST_PAD_SINK) { + gint fac = (fp) ? 4 : 2; + + /* asked to convert size of an incoming buffer */ + size /= fac; + *othersize = (size * ratio_den + (ratio_num >> 1)) / ratio_num; + *othersize *= fac; + } else { + gint fac = (fp) ? 4 : 2; + + /* asked to convert size of an outgoing buffer */ + size /= fac; + *othersize = (size * ratio_num + (ratio_den >> 1)) / ratio_den; + *othersize *= fac; + } + + GST_LOG ("transformed size %d to %d", size, *othersize); + + if (!use_internal) + resample_resampler_destroy (state); + + return ret; +} + +static gboolean +gst_speex_resample_set_caps (GstBaseTransform * base, GstCaps * incaps, + GstCaps * outcaps) +{ + gboolean ret; + gint inrate = 0, outrate = 0, channels = 0; + gboolean fp = FALSE; + GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); + + GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %" + GST_PTR_FORMAT, incaps, outcaps); + + ret = gst_speex_resample_parse_caps (incaps, outcaps, + &channels, &inrate, &outrate, &fp); + + g_return_val_if_fail (ret, FALSE); + + ret = + gst_speex_resample_update_state (resample, channels, inrate, outrate, + resample->quality, fp); + + g_return_val_if_fail (ret, FALSE); + + /* save caps so we can short-circuit in the size_transform if the caps + * are the same */ + gst_caps_replace (&resample->sinkcaps, incaps); + gst_caps_replace (&resample->srccaps, outcaps); + + return TRUE; +} + +static gboolean +gst_speex_resample_event (GstBaseTransform * base, GstEvent * event) +{ + GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_START: + break; + case GST_EVENT_FLUSH_STOP: + case GST_EVENT_NEWSEGMENT: + gst_speex_resample_reset_state (resample); + resample->ts_offset = -1; + resample->next_ts = -1; + resample->offset = -1; + break; + case GST_EVENT_EOS: + gst_speex_resample_reset_state (resample); + break; + default: + break; + } + parent_class->event (base, event); + + return TRUE; +} + +static gboolean +gst_speex_resample_check_discont (GstSpeexResample * resample, + GstClockTime timestamp) +{ + if (timestamp != GST_CLOCK_TIME_NONE && + resample->prev_ts != GST_CLOCK_TIME_NONE && + resample->prev_duration != GST_CLOCK_TIME_NONE && + timestamp != resample->prev_ts + resample->prev_duration) { + /* Potentially a discontinuous buffer. However, it turns out that many + * elements generate imperfect streams due to rounding errors, so we permit + * a small error (up to one sample) without triggering a filter + * flush/restart (if triggered incorrectly, this will be audible) */ + GstClockTimeDiff diff = timestamp - + (resample->prev_ts + resample->prev_duration); + + if (ABS (diff) > GST_SECOND / resample->inrate) { + GST_WARNING ("encountered timestamp discontinuity of %" G_GINT64_FORMAT, + diff); + return TRUE; + } + } + + return FALSE; +} + +static GstFlowReturn +gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf, + GstBuffer * outbuf) +{ + guint32 in_len, in_processed; + guint32 out_len, out_processed; + gint err = RESAMPLER_ERR_SUCCESS; + + in_len = GST_BUFFER_SIZE (inbuf) / resample->channels; + out_len = GST_BUFFER_SIZE (outbuf) / resample->channels; + + if (resample->fp) { + in_len /= 4; + out_len /= 4; + } else { + in_len /= 2; + out_len /= 2; + } + + in_processed = in_len; + out_processed = out_len; + + if (resample->fp) + err = resample_float_resampler_process_interleaved_float (resample->state, + (const gfloat *) GST_BUFFER_DATA (inbuf), &in_processed, + (gfloat *) GST_BUFFER_DATA (outbuf), &out_processed); + else + err = resample_int_resampler_process_interleaved_int (resample->state, + (const gint16 *) GST_BUFFER_DATA (inbuf), &in_processed, + (gint16 *) GST_BUFFER_DATA (outbuf), &out_processed); + + if (in_len != in_processed) + GST_WARNING ("Converted %d of %d input samples", in_processed, in_len); + + if (out_len != out_processed) + GST_WARNING ("Converted to %d instead of %d output samples", out_processed, + out_len); + + if (err != RESAMPLER_ERR_SUCCESS) { + GST_ERROR ("Failed to convert data: %s", resample_resampler_strerror (err)); + return GST_FLOW_ERROR; + } else { + return GST_FLOW_OK; + } +} + +static GstFlowReturn +gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf, + GstBuffer * outbuf) +{ + GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base); + guint8 *data; + gulong size; + GstClockTime timestamp; + gint outsamples; + + if (resample->state == NULL) + if (!(resample->state = gst_speex_resample_init_state (resample->channels, + resample->inrate, resample->outrate, resample->quality, + resample->fp))) + return GST_FLOW_ERROR; + + data = GST_BUFFER_DATA (inbuf); + size = GST_BUFFER_SIZE (inbuf); + timestamp = GST_BUFFER_TIMESTAMP (inbuf); + + GST_LOG ("transforming buffer of %ld bytes, ts %" + GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %" + G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT, + size, GST_TIME_ARGS (timestamp), + GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)), + GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf)); + + /* check for timestamp discontinuities and flush/reset if needed */ + if (G_UNLIKELY (gst_speex_resample_check_discont (resample, timestamp) + || GST_BUFFER_IS_DISCONT (inbuf))) { + /* Flush internal samples */ + gst_speex_resample_reset_state (resample); + /* Inform downstream element about discontinuity */ + resample->need_discont = TRUE; + /* We want to recalculate the offset */ + resample->ts_offset = -1; + } + + outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels; + outsamples /= (resample->fp) ? 4 : 2; + + if (resample->ts_offset == -1) { + /* if we don't know the initial offset yet, calculate it based on the + * input timestamp. */ + if (GST_CLOCK_TIME_IS_VALID (timestamp)) { + GstClockTime stime; + + /* offset used to calculate the timestamps. We use the sample offset for + * this to make it more accurate. We want the first buffer to have the + * same timestamp as the incoming timestamp. */ + resample->next_ts = timestamp; + resample->ts_offset = + gst_util_uint64_scale_int (timestamp, resample->outrate, GST_SECOND); + /* offset used to set as the buffer offset, this offset is always + * relative to the stream time, note that timestamp is not... */ + stime = (timestamp - base->segment.start) + base->segment.time; + resample->offset = + gst_util_uint64_scale_int (stime, resample->outrate, GST_SECOND); + } + } + resample->prev_ts = timestamp; + resample->prev_duration = GST_BUFFER_DURATION (inbuf); + + GST_BUFFER_OFFSET (outbuf) = resample->offset; + GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts; + + if (resample->ts_offset != -1) { + resample->offset += outsamples; + resample->ts_offset += outsamples; + resample->next_ts = + gst_util_uint64_scale_int (resample->ts_offset, GST_SECOND, + resample->outrate); + GST_BUFFER_OFFSET_END (outbuf) = resample->offset; + + /* we calculate DURATION as the difference between "next" timestamp + * and current timestamp so we ensure a contiguous stream, instead of + * having rounding errors. */ + GST_BUFFER_DURATION (outbuf) = resample->next_ts - + GST_BUFFER_TIMESTAMP (outbuf); + } else { + /* no valid offset know, we can still sortof calculate the duration though */ + GST_BUFFER_DURATION (outbuf) = + gst_util_uint64_scale_int (outsamples, GST_SECOND, resample->outrate); + } + + if (G_UNLIKELY (resample->need_discont)) { + GST_DEBUG ("marking this buffer with the DISCONT flag"); + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); + resample->need_discont = FALSE; + } + + return gst_speex_resample_process (resample, inbuf, outbuf); +} + +static void +gst_speex_resample_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstSpeexResample *resample; + + resample = GST_SPEEX_RESAMPLE (object); + + switch (prop_id) { + case PROP_QUALITY: + resample->quality = g_value_get_int (value); + GST_DEBUG ("new quality %d", resample->quality); + + gst_speex_resample_update_state (resample, resample->channels, + resample->inrate, resample->outrate, resample->quality, resample->fp); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_speex_resample_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstSpeexResample *resample; + + resample = GST_SPEEX_RESAMPLE (object); + + switch (prop_id) { + case PROP_QUALITY: + g_value_set_int (value, resample->quality); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + + +static gboolean +plugin_init (GstPlugin * plugin) +{ + if (!gst_element_register (plugin, "speexresample", GST_RANK_NONE, + GST_TYPE_SPEEX_RESAMPLE)) { + return FALSE; + } + + return TRUE; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "speexresample", + "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, + GST_PACKAGE_ORIGIN); diff --git a/gst/speexresample/gstspeexresample.h b/gst/speexresample/gstspeexresample.h new file mode 100644 index 00000000..68731289 --- /dev/null +++ b/gst/speexresample/gstspeexresample.h @@ -0,0 +1,80 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * Copyright (C) <2007> Sebastian Dröge <slomo@circular-chaos.org> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __SPEEX_RESAMPLE_H__ +#define __SPEEX_RESAMPLE_H__ + +#include <gst/gst.h> +#include <gst/base/gstbasetransform.h> + +#include "speex_resampler_wrapper.h" + +G_BEGIN_DECLS + +#define GST_TYPE_SPEEX_RESAMPLE \ + (gst_speex_resample_get_type()) +#define GST_SPEEX_RESAMPLE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResample)) +#define GST_SPEEX_RESAMPLE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResampleClass)) +#define GST_IS_SPEEX_RESAMPLE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_RESAMPLE)) +#define GST_IS_SPEEX_RESAMPLE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_RESAMPLE)) + +typedef struct _GstSpeexResample GstSpeexResample; +typedef struct _GstSpeexResampleClass GstSpeexResampleClass; + +/** + * GstSpeexResample: + * + * Opaque data structure. + */ +struct _GstSpeexResample { + GstBaseTransform element; + + GstCaps *srccaps, *sinkcaps; + + gboolean need_discont; + + guint64 offset; + guint64 ts_offset; + GstClockTime next_ts; + GstClockTime prev_ts, prev_duration; + + gboolean fp; + int channels; + int inrate; + int outrate; + int quality; + + SpeexResamplerState *state; +}; + +struct _GstSpeexResampleClass { + GstBaseTransformClass parent_class; +}; + +GType gst_speex_resample_get_type(void); + +G_END_DECLS + +#endif /* __SPEEX_RESAMPLE_H__ */ diff --git a/gst/speexresample/resample.c b/gst/speexresample/resample.c new file mode 100644 index 00000000..f3c97fdd --- /dev/null +++ b/gst/speexresample/resample.c @@ -0,0 +1,1310 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: resample.c + Arbitrary resampling code + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +/* + The design goals of this code are: + - Very fast algorithm + - SIMD-friendly algorithm + - Low memory requirement + - Good *perceptual* quality (and not best SNR) + + Warning: This resampler is relatively new. Although I think I got rid of + all the major bugs and I don't expect the API to change anymore, there + may be something I've missed. So use with caution. + + This algorithm is based on this original resampling algorithm: + Smith, Julius O. Digital Audio Resampling Home Page + Center for Computer Research in Music and Acoustics (CCRMA), + Stanford University, 2007. + Web published at http://www-ccrma.stanford.edu/~jos/resample/. + + There is one main difference, though. This resampler uses cubic + interpolation instead of linear interpolation in the above paper. This + makes the table much smaller and makes it possible to compute that table + on a per-stream basis. In turn, being able to tweak the table for each + stream makes it possible to both reduce complexity on simple ratios + (e.g. 2/3), and get rid of the rounding operations in the inner loop. + The latter both reduces CPU time and makes the algorithm more SIMD-friendly. +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#ifdef OUTSIDE_SPEEX +#include <stdlib.h> +static void * +speex_alloc (int size) +{ + return calloc (size, 1); +} +static void * +speex_realloc (void *ptr, int size) +{ + return realloc (ptr, size); +} +static void +speex_free (void *ptr) +{ + free (ptr); +} + +#include "speex_resampler.h" +#include "arch.h" +#else /* OUTSIDE_SPEEX */ + +#include "speex/speex_resampler.h" +#include "arch.h" +#include "os_support.h" +#endif /* OUTSIDE_SPEEX */ + +#include <math.h> + +#ifndef M_PI +#define M_PI 3.14159263 +#endif + +#ifdef FIXED_POINT +#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x))) +#else +#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x)))) +#endif + +/*#define float double*/ +#define FILTER_SIZE 64 +#define OVERSAMPLE 8 + +#define IMAX(a,b) ((a) > (b) ? (a) : (b)) +#define IMIN(a,b) ((a) < (b) ? (a) : (b)) + +#ifndef NULL +#define NULL 0 +#endif + +typedef int (*resampler_basic_func) (SpeexResamplerState *, spx_uint32_t, + const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *); + +struct SpeexResamplerState_ +{ + spx_uint32_t in_rate; + spx_uint32_t out_rate; + spx_uint32_t num_rate; + spx_uint32_t den_rate; + + int quality; + spx_uint32_t nb_channels; + spx_uint32_t filt_len; + spx_uint32_t mem_alloc_size; + int int_advance; + int frac_advance; + float cutoff; + spx_uint32_t oversample; + int initialised; + int started; + + /* These are per-channel */ + spx_int32_t *last_sample; + spx_uint32_t *samp_frac_num; + spx_uint32_t *magic_samples; + + spx_word16_t *mem; + spx_word16_t *sinc_table; + spx_uint32_t sinc_table_length; + resampler_basic_func resampler_ptr; + + int in_stride; + int out_stride; +}; + +static double kaiser12_table[68] = { + 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076, + 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014, + 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601, + 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014, + 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490, + 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546, + 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178, + 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947, + 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058, + 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438, + 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734, + 0.00001000, 0.00000000 +}; + +/* +static double kaiser12_table[36] = { + 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741, + 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762, + 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274, + 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466, + 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291, + 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000}; +*/ +static double kaiser10_table[36] = { + 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446, + 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347, + 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962, + 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451, + 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739, + 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000 +}; + +static double kaiser8_table[36] = { + 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200, + 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126, + 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272, + 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758, + 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490, + 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000 +}; + +static double kaiser6_table[36] = { + 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003, + 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565, + 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561, + 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058, + 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600, + 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000 +}; + +struct FuncDef +{ + double *table; + int oversample; +}; + +static struct FuncDef _KAISER12 = { kaiser12_table, 64 }; + +#define KAISER12 (&_KAISER12) +/*static struct FuncDef _KAISER12 = {kaiser12_table, 32}; +#define KAISER12 (&_KAISER12)*/ +static struct FuncDef _KAISER10 = { kaiser10_table, 32 }; + +#define KAISER10 (&_KAISER10) +static struct FuncDef _KAISER8 = { kaiser8_table, 32 }; + +#define KAISER8 (&_KAISER8) +static struct FuncDef _KAISER6 = { kaiser6_table, 32 }; + +#define KAISER6 (&_KAISER6) + +struct QualityMapping +{ + int base_length; + int oversample; + float downsample_bandwidth; + float upsample_bandwidth; + struct FuncDef *window_func; +}; + + +/* This table maps conversion quality to internal parameters. There are two + reasons that explain why the up-sampling bandwidth is larger than the + down-sampling bandwidth: + 1) When up-sampling, we can assume that the spectrum is already attenuated + close to the Nyquist rate (from an A/D or a previous resampling filter) + 2) Any aliasing that occurs very close to the Nyquist rate will be masked + by the sinusoids/noise just below the Nyquist rate (guaranteed only for + up-sampling). +*/ +static const struct QualityMapping quality_map[11] = { + {8, 4, 0.830f, 0.860f, KAISER6}, /* Q0 */ + {16, 4, 0.850f, 0.880f, KAISER6}, /* Q1 */ + {32, 4, 0.882f, 0.910f, KAISER6}, /* Q2 *//* 82.3% cutoff ( ~60 dB stop) 6 */ + {48, 8, 0.895f, 0.917f, KAISER8}, /* Q3 *//* 84.9% cutoff ( ~80 dB stop) 8 */ + {64, 8, 0.921f, 0.940f, KAISER8}, /* Q4 *//* 88.7% cutoff ( ~80 dB stop) 8 */ + {80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 *//* 89.1% cutoff (~100 dB stop) 10 */ + {96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 *//* 91.5% cutoff (~100 dB stop) 10 */ + {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 *//* 93.1% cutoff (~100 dB stop) 10 */ + {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 *//* 94.5% cutoff (~100 dB stop) 10 */ + {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 *//* 95.5% cutoff (~100 dB stop) 10 */ + {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 *//* 96.6% cutoff (~100 dB stop) 10 */ +}; + +/*8,24,40,56,80,104,128,160,200,256,320*/ +static double +compute_func (float x, struct FuncDef *func) +{ + float y, frac; + double interp[4]; + int ind; + + y = x * func->oversample; + ind = (int) floor (y); + frac = (y - ind); + /* CSE with handle the repeated powers */ + interp[3] = -0.1666666667 * frac + 0.1666666667 * (frac * frac * frac); + interp[2] = frac + 0.5 * (frac * frac) - 0.5 * (frac * frac * frac); + /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */ + interp[0] = + -0.3333333333 * frac + 0.5 * (frac * frac) - + 0.1666666667 * (frac * frac * frac); + /* Just to make sure we don't have rounding problems */ + interp[1] = 1.f - interp[3] - interp[2] - interp[0]; + + /*sum = frac*accum[1] + (1-frac)*accum[2]; */ + return interp[0] * func->table[ind] + interp[1] * func->table[ind + 1] + + interp[2] * func->table[ind + 2] + interp[3] * func->table[ind + 3]; +} + +#if 0 +#include <stdio.h> +int +main (int argc, char **argv) +{ + int i; + + for (i = 0; i < 256; i++) { + printf ("%f\n", compute_func (i / 256., KAISER12)); + } + return 0; +} +#endif + +#ifdef FIXED_POINT +/* The slow way of computing a sinc for the table. Should improve that some day */ +static spx_word16_t +sinc (float cutoff, float x, int N, struct FuncDef *window_func) +{ + /*fprintf (stderr, "%f ", x); */ + float xx = x * cutoff; + + if (fabs (x) < 1e-6f) + return WORD2INT (32768. * cutoff); + else if (fabs (x) > .5f * N) + return 0; + /*FIXME: Can it really be any slower than this? */ + return WORD2INT (32768. * cutoff * sin (M_PI * xx) / (M_PI * xx) * + compute_func (fabs (2. * x / N), window_func)); +} +#else +/* The slow way of computing a sinc for the table. Should improve that some day */ +static spx_word16_t +sinc (float cutoff, float x, int N, struct FuncDef *window_func) +{ + /*fprintf (stderr, "%f ", x); */ + float xx = x * cutoff; + + if (fabs (x) < 1e-6) + return cutoff; + else if (fabs (x) > .5 * N) + return 0; + /*FIXME: Can it really be any slower than this? */ + return cutoff * sin (M_PI * xx) / (M_PI * xx) * compute_func (fabs (2. * x / + N), window_func); +} +#endif + +#ifdef FIXED_POINT +static void +cubic_coef (spx_word16_t x, spx_word16_t interp[4]) +{ + /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation + but I know it's MMSE-optimal on a sinc */ + spx_word16_t x2, x3; + + x2 = MULT16_16_P15 (x, x); + x3 = MULT16_16_P15 (x, x2); + interp[0] = + PSHR32 (MULT16_16 (QCONST16 (-0.16667f, 15), + x) + MULT16_16 (QCONST16 (0.16667f, 15), x3), 15); + interp[1] = + EXTRACT16 (EXTEND32 (x) + SHR32 (SUB32 (EXTEND32 (x2), EXTEND32 (x3)), + 1)); + interp[3] = + PSHR32 (MULT16_16 (QCONST16 (-0.33333f, 15), + x) + MULT16_16 (QCONST16 (.5f, 15), + x2) - MULT16_16 (QCONST16 (0.16667f, 15), x3), 15); + /* Just to make sure we don't have rounding problems */ + interp[2] = Q15_ONE - interp[0] - interp[1] - interp[3]; + if (interp[2] < 32767) + interp[2] += 1; +} +#else +static void +cubic_coef (spx_word16_t frac, spx_word16_t interp[4]) +{ + /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation + but I know it's MMSE-optimal on a sinc */ + interp[0] = -0.16667f * frac + 0.16667f * frac * frac * frac; + interp[1] = frac + 0.5f * frac * frac - 0.5f * frac * frac * frac; + /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */ + interp[3] = + -0.33333f * frac + 0.5f * frac * frac - 0.16667f * frac * frac * frac; + /* Just to make sure we don't have rounding problems */ + interp[2] = 1. - interp[0] - interp[1] - interp[3]; +} +#endif + +static int +resampler_basic_direct_single (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len, + spx_word16_t * out, spx_uint32_t * out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= (spx_int32_t) * in_len + || out_sample >= (spx_int32_t) * out_len)) { + int j; + spx_word32_t sum = 0; + + /* We already have all the filter coefficients pre-computed in the table */ + const spx_word16_t *ptr; + + /* Do the memory part */ + for (j = 0; last_sample - N + 1 + j < 0; j++) { + sum += + MULT16_16 (mem[last_sample + j], + st->sinc_table[samp_frac_num * st->filt_len + j]); + } + + /* Do the new part */ + ptr = in + st->in_stride * (last_sample - N + 1 + j); + for (; j < N; j++) { + sum += MULT16_16 (*ptr, st->sinc_table[samp_frac_num * st->filt_len + j]); + ptr += st->in_stride; + } + + *out = PSHR32 (sum, 15); + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) { + samp_frac_num -= st->den_rate; + last_sample++; + } + } + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +#ifdef FIXED_POINT +#else +/* This is the same as the previous function, except with a double-precision accumulator */ +static int +resampler_basic_direct_double (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len, + spx_word16_t * out, spx_uint32_t * out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= (spx_int32_t) * in_len + || out_sample >= (spx_int32_t) * out_len)) { + int j; + double sum = 0; + + /* We already have all the filter coefficients pre-computed in the table */ + const spx_word16_t *ptr; + + /* Do the memory part */ + for (j = 0; last_sample - N + 1 + j < 0; j++) { + sum += + MULT16_16 (mem[last_sample + j], + (double) st->sinc_table[samp_frac_num * st->filt_len + j]); + } + + /* Do the new part */ + ptr = in + st->in_stride * (last_sample - N + 1 + j); + for (; j < N; j++) { + sum += + MULT16_16 (*ptr, + (double) st->sinc_table[samp_frac_num * st->filt_len + j]); + ptr += st->in_stride; + } + + *out = sum; + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) { + samp_frac_num -= st->den_rate; + last_sample++; + } + } + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} +#endif + +static int +resampler_basic_interpolate_single (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len, + spx_word16_t * out, spx_uint32_t * out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= (spx_int32_t) * in_len + || out_sample >= (spx_int32_t) * out_len)) { + int j; + spx_word32_t sum = 0; + + /* We need to interpolate the sinc filter */ + spx_word32_t accum[4] = { 0.f, 0.f, 0.f, 0.f }; + spx_word16_t interp[4]; + const spx_word16_t *ptr; + int offset; + spx_word16_t frac; + + offset = samp_frac_num * st->oversample / st->den_rate; +#ifdef FIXED_POINT + frac = + PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15), + st->den_rate); +#else + frac = + ((float) ((samp_frac_num * st->oversample) % st->den_rate)) / + st->den_rate; +#endif + /* This code is written like this to make it easy to optimise with SIMD. + For most DSPs, it would be best to split the loops in two because most DSPs + have only two accumulators */ + for (j = 0; last_sample - N + 1 + j < 0; j++) { + spx_word16_t curr_mem = mem[last_sample + j]; + + accum[0] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]); + accum[1] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]); + accum[2] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset]); + accum[3] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]); + } + ptr = in + st->in_stride * (last_sample - N + 1 + j); + /* Do the new part */ + for (; j < N; j++) { + spx_word16_t curr_in = *ptr; + + ptr += st->in_stride; + accum[0] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]); + accum[1] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]); + accum[2] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset]); + accum[3] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]); + } + cubic_coef (frac, interp); + sum = + MULT16_32_Q15 (interp[0], accum[0]) + MULT16_32_Q15 (interp[1], + accum[1]) + MULT16_32_Q15 (interp[2], + accum[2]) + MULT16_32_Q15 (interp[3], accum[3]); + + *out = PSHR32 (sum, 15); + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) { + samp_frac_num -= st->den_rate; + last_sample++; + } + } + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +#ifdef FIXED_POINT +#else +/* This is the same as the previous function, except with a double-precision accumulator */ +static int +resampler_basic_interpolate_double (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len, + spx_word16_t * out, spx_uint32_t * out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; + + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= (spx_int32_t) * in_len + || out_sample >= (spx_int32_t) * out_len)) { + int j; + spx_word32_t sum = 0; + + /* We need to interpolate the sinc filter */ + double accum[4] = { 0.f, 0.f, 0.f, 0.f }; + float interp[4]; + const spx_word16_t *ptr; + float alpha = ((float) samp_frac_num) / st->den_rate; + int offset = samp_frac_num * st->oversample / st->den_rate; + float frac = alpha * st->oversample - offset; + + /* This code is written like this to make it easy to optimise with SIMD. + For most DSPs, it would be best to split the loops in two because most DSPs + have only two accumulators */ + for (j = 0; last_sample - N + 1 + j < 0; j++) { + double curr_mem = mem[last_sample + j]; + + accum[0] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]); + accum[1] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]); + accum[2] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset]); + accum[3] += + MULT16_16 (curr_mem, + st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]); + } + ptr = in + st->in_stride * (last_sample - N + 1 + j); + /* Do the new part */ + for (; j < N; j++) { + double curr_in = *ptr; + + ptr += st->in_stride; + accum[0] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]); + accum[1] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]); + accum[2] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset]); + accum[3] += + MULT16_16 (curr_in, + st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]); + } + cubic_coef (frac, interp); + sum = + interp[0] * accum[0] + interp[1] * accum[1] + interp[2] * accum[2] + + interp[3] * accum[3]; + + *out = PSHR32 (sum, 15); + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) { + samp_frac_num -= st->den_rate; + last_sample++; + } + } + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} +#endif + +static void +update_filter (SpeexResamplerState * st) +{ + spx_uint32_t old_length; + + old_length = st->filt_len; + st->oversample = quality_map[st->quality].oversample; + st->filt_len = quality_map[st->quality].base_length; + + if (st->num_rate > st->den_rate) { + /* down-sampling */ + st->cutoff = + quality_map[st->quality].downsample_bandwidth * st->den_rate / + st->num_rate; + /* FIXME: divide the numerator and denominator by a certain amount if they're too large */ + st->filt_len = st->filt_len * st->num_rate / st->den_rate; + /* Round down to make sure we have a multiple of 4 */ + st->filt_len &= (~0x3); + if (2 * st->den_rate < st->num_rate) + st->oversample >>= 1; + if (4 * st->den_rate < st->num_rate) + st->oversample >>= 1; + if (8 * st->den_rate < st->num_rate) + st->oversample >>= 1; + if (16 * st->den_rate < st->num_rate) + st->oversample >>= 1; + if (st->oversample < 1) + st->oversample = 1; + } else { + /* up-sampling */ + st->cutoff = quality_map[st->quality].upsample_bandwidth; + } + + /* Choose the resampling type that requires the least amount of memory */ + if (st->den_rate <= st->oversample) { + spx_uint32_t i; + + if (!st->sinc_table) + st->sinc_table = + (spx_word16_t *) speex_alloc (st->filt_len * st->den_rate * + sizeof (spx_word16_t)); + else if (st->sinc_table_length < st->filt_len * st->den_rate) { + st->sinc_table = + (spx_word16_t *) speex_realloc (st->sinc_table, + st->filt_len * st->den_rate * sizeof (spx_word16_t)); + st->sinc_table_length = st->filt_len * st->den_rate; + } + for (i = 0; i < st->den_rate; i++) { + spx_int32_t j; + + for (j = 0; j < st->filt_len; j++) { + st->sinc_table[i * st->filt_len + j] = + sinc (st->cutoff, + ((j - (spx_int32_t) st->filt_len / 2 + 1) - + ((float) i) / st->den_rate), st->filt_len, + quality_map[st->quality].window_func); + } + } +#ifdef FIXED_POINT + st->resampler_ptr = resampler_basic_direct_single; +#else + if (st->quality > 8) + st->resampler_ptr = resampler_basic_direct_double; + else + st->resampler_ptr = resampler_basic_direct_single; +#endif + /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff); */ + } else { + spx_int32_t i; + + if (!st->sinc_table) + st->sinc_table = + (spx_word16_t *) speex_alloc ((st->filt_len * st->oversample + + 8) * sizeof (spx_word16_t)); + else if (st->sinc_table_length < st->filt_len * st->oversample + 8) { + st->sinc_table = + (spx_word16_t *) speex_realloc (st->sinc_table, + (st->filt_len * st->oversample + 8) * sizeof (spx_word16_t)); + st->sinc_table_length = st->filt_len * st->oversample + 8; + } + for (i = -4; i < (spx_int32_t) (st->oversample * st->filt_len + 4); i++) + st->sinc_table[i + 4] = + sinc (st->cutoff, (i / (float) st->oversample - st->filt_len / 2), + st->filt_len, quality_map[st->quality].window_func); +#ifdef FIXED_POINT + st->resampler_ptr = resampler_basic_interpolate_single; +#else + if (st->quality > 8) + st->resampler_ptr = resampler_basic_interpolate_double; + else + st->resampler_ptr = resampler_basic_interpolate_single; +#endif + /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff); */ + } + st->int_advance = st->num_rate / st->den_rate; + st->frac_advance = st->num_rate % st->den_rate; + + + /* Here's the place where we update the filter memory to take into account + the change in filter length. It's probably the messiest part of the code + due to handling of lots of corner cases. */ + if (!st->mem) { + spx_uint32_t i; + + st->mem = + (spx_word16_t *) speex_alloc (st->nb_channels * (st->filt_len - + 1) * sizeof (spx_word16_t)); + for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++) + st->mem[i] = 0; + st->mem_alloc_size = st->filt_len - 1; + /*speex_warning("init filter"); */ + } else if (!st->started) { + spx_uint32_t i; + + st->mem = + (spx_word16_t *) speex_realloc (st->mem, + st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t)); + for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++) + st->mem[i] = 0; + st->mem_alloc_size = st->filt_len - 1; + /*speex_warning("reinit filter"); */ + } else if (st->filt_len > old_length) { + spx_int32_t i; + + /* Increase the filter length */ + /*speex_warning("increase filter size"); */ + int old_alloc_size = st->mem_alloc_size; + + if (st->filt_len - 1 > st->mem_alloc_size) { + st->mem = + (spx_word16_t *) speex_realloc (st->mem, + st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t)); + st->mem_alloc_size = st->filt_len - 1; + } + for (i = st->nb_channels - 1; i >= 0; i--) { + spx_int32_t j; + spx_uint32_t olen = old_length; + + /*if (st->magic_samples[i]) */ + { + /* Try and remove the magic samples as if nothing had happened */ + + /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */ + olen = old_length + 2 * st->magic_samples[i]; + for (j = old_length - 2 + st->magic_samples[i]; j >= 0; j--) + st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]] = + st->mem[i * old_alloc_size + j]; + for (j = 0; j < st->magic_samples[i]; j++) + st->mem[i * st->mem_alloc_size + j] = 0; + st->magic_samples[i] = 0; + } + if (st->filt_len > olen) { + /* If the new filter length is still bigger than the "augmented" length */ + /* Copy data going backward */ + for (j = 0; j < olen - 1; j++) + st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] = + st->mem[i * st->mem_alloc_size + (olen - 2 - j)]; + /* Then put zeros for lack of anything better */ + for (; j < st->filt_len - 1; j++) + st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] = 0; + /* Adjust last_sample */ + st->last_sample[i] += (st->filt_len - olen) / 2; + } else { + /* Put back some of the magic! */ + st->magic_samples[i] = (olen - st->filt_len) / 2; + for (j = 0; j < st->filt_len - 1 + st->magic_samples[i]; j++) + st->mem[i * st->mem_alloc_size + j] = + st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]]; + } + } + } else if (st->filt_len < old_length) { + spx_uint32_t i; + + /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic" + samples so they can be used directly as input the next time(s) */ + for (i = 0; i < st->nb_channels; i++) { + spx_uint32_t j; + spx_uint32_t old_magic = st->magic_samples[i]; + + st->magic_samples[i] = (old_length - st->filt_len) / 2; + /* We must copy some of the memory that's no longer used */ + /* Copy data going backward */ + for (j = 0; j < st->filt_len - 1 + st->magic_samples[i] + old_magic; j++) + st->mem[i * st->mem_alloc_size + j] = + st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]]; + st->magic_samples[i] += old_magic; + } + } + +} + +SpeexResamplerState * +speex_resampler_init (spx_uint32_t nb_channels, spx_uint32_t in_rate, + spx_uint32_t out_rate, int quality, int *err) +{ + return speex_resampler_init_frac (nb_channels, in_rate, out_rate, in_rate, + out_rate, quality, err); +} + +SpeexResamplerState * +speex_resampler_init_frac (spx_uint32_t nb_channels, spx_uint32_t ratio_num, + spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, + int quality, int *err) +{ + spx_uint32_t i; + SpeexResamplerState *st; + + if (quality > 10 || quality < 0) { + if (err) + *err = RESAMPLER_ERR_INVALID_ARG; + return NULL; + } + st = (SpeexResamplerState *) speex_alloc (sizeof (SpeexResamplerState)); + st->initialised = 0; + st->started = 0; + st->in_rate = 0; + st->out_rate = 0; + st->num_rate = 0; + st->den_rate = 0; + st->quality = -1; + st->sinc_table_length = 0; + st->mem_alloc_size = 0; + st->filt_len = 0; + st->mem = 0; + st->resampler_ptr = 0; + + st->cutoff = 1.f; + st->nb_channels = nb_channels; + st->in_stride = 1; + st->out_stride = 1; + + /* Per channel data */ + st->last_sample = (spx_int32_t *) speex_alloc (nb_channels * sizeof (int)); + st->magic_samples = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int)); + st->samp_frac_num = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int)); + for (i = 0; i < nb_channels; i++) { + st->last_sample[i] = 0; + st->magic_samples[i] = 0; + st->samp_frac_num[i] = 0; + } + + speex_resampler_set_quality (st, quality); + speex_resampler_set_rate_frac (st, ratio_num, ratio_den, in_rate, out_rate); + + + update_filter (st); + + st->initialised = 1; + if (err) + *err = RESAMPLER_ERR_SUCCESS; + + return st; +} + +void +speex_resampler_destroy (SpeexResamplerState * st) +{ + speex_free (st->mem); + speex_free (st->sinc_table); + speex_free (st->last_sample); + speex_free (st->magic_samples); + speex_free (st->samp_frac_num); + speex_free (st); +} + + + +static int +speex_resampler_process_native (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len, + spx_word16_t * out, spx_uint32_t * out_len) +{ + int j = 0; + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + spx_uint32_t tmp_out_len = 0; + + mem = st->mem + channel_index * st->mem_alloc_size; + st->started = 1; + + /* Handle the case where we have samples left from a reduction in filter length */ + if (st->magic_samples[channel_index]) { + int istride_save; + spx_uint32_t tmp_in_len; + spx_uint32_t tmp_magic; + + istride_save = st->in_stride; + tmp_in_len = st->magic_samples[channel_index]; + tmp_out_len = *out_len; + /* magic_samples needs to be set to zero to avoid infinite recursion */ + tmp_magic = st->magic_samples[channel_index]; + st->magic_samples[channel_index] = 0; + st->in_stride = 1; + speex_resampler_process_native (st, channel_index, mem + N - 1, &tmp_in_len, + out, &tmp_out_len); + st->in_stride = istride_save; + /*speex_warning_int("extra samples:", tmp_out_len); */ + /* If we couldn't process all "magic" input samples, save the rest for next time */ + if (tmp_in_len < tmp_magic) { + spx_uint32_t i; + + st->magic_samples[channel_index] = tmp_magic - tmp_in_len; + for (i = 0; i < st->magic_samples[channel_index]; i++) + mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len]; + } + out += tmp_out_len * st->out_stride; + *out_len -= tmp_out_len; + } + + /* Call the right resampler through the function ptr */ + out_sample = st->resampler_ptr (st, channel_index, in, in_len, out, out_len); + + if (st->last_sample[channel_index] < (spx_int32_t) * in_len) + *in_len = st->last_sample[channel_index]; + *out_len = out_sample + tmp_out_len; + st->last_sample[channel_index] -= *in_len; + + for (j = 0; j < N - 1 - (spx_int32_t) * in_len; j++) + mem[j] = mem[j + *in_len]; + for (; j < N - 1; j++) + mem[j] = in[st->in_stride * (j + *in_len - N + 1)]; + + return RESAMPLER_ERR_SUCCESS; +} + +#define FIXED_STACK_ALLOC 1024 + +#ifdef FIXED_POINT +int +speex_resampler_process_float (SpeexResamplerState * st, + spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len, + float *out, spx_uint32_t * out_len) +{ + spx_uint32_t i; + int istride_save, ostride_save; + +#ifdef VAR_ARRAYS + spx_word16_t x[*in_len]; + spx_word16_t y[*out_len]; + + /*VARDECL(spx_word16_t *x); + VARDECL(spx_word16_t *y); + ALLOC(x, *in_len, spx_word16_t); + ALLOC(y, *out_len, spx_word16_t); */ + istride_save = st->in_stride; + ostride_save = st->out_stride; + for (i = 0; i < *in_len; i++) + x[i] = WORD2INT (in[i * st->in_stride]); + st->in_stride = st->out_stride = 1; + speex_resampler_process_native (st, channel_index, x, in_len, y, out_len); + st->in_stride = istride_save; + st->out_stride = ostride_save; + for (i = 0; i < *out_len; i++) + out[i * st->out_stride] = y[i]; +#else + spx_word16_t x[FIXED_STACK_ALLOC]; + spx_word16_t y[FIXED_STACK_ALLOC]; + spx_uint32_t ilen = *in_len, olen = *out_len; + + istride_save = st->in_stride; + ostride_save = st->out_stride; + while (ilen && olen) { + spx_uint32_t ichunk, ochunk; + + ichunk = ilen; + ochunk = olen; + if (ichunk > FIXED_STACK_ALLOC) + ichunk = FIXED_STACK_ALLOC; + if (ochunk > FIXED_STACK_ALLOC) + ochunk = FIXED_STACK_ALLOC; + for (i = 0; i < ichunk; i++) + x[i] = WORD2INT (in[i * st->in_stride]); + st->in_stride = st->out_stride = 1; + speex_resampler_process_native (st, channel_index, x, &ichunk, y, &ochunk); + st->in_stride = istride_save; + st->out_stride = ostride_save; + for (i = 0; i < ochunk; i++) + out[i * st->out_stride] = y[i]; + out += ochunk; + in += ichunk; + ilen -= ichunk; + olen -= ochunk; + } + *in_len -= ilen; + *out_len -= olen; +#endif + return RESAMPLER_ERR_SUCCESS; +} + +int +speex_resampler_process_int (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len, + spx_int16_t * out, spx_uint32_t * out_len) +{ + return speex_resampler_process_native (st, channel_index, in, in_len, out, + out_len); +} +#else +int +speex_resampler_process_float (SpeexResamplerState * st, + spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len, + float *out, spx_uint32_t * out_len) +{ + return speex_resampler_process_native (st, channel_index, in, in_len, out, + out_len); +} + +int +speex_resampler_process_int (SpeexResamplerState * st, + spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len, + spx_int16_t * out, spx_uint32_t * out_len) +{ + spx_uint32_t i; + int istride_save, ostride_save; + +#ifdef VAR_ARRAYS + spx_word16_t x[*in_len]; + spx_word16_t y[*out_len]; + + /*VARDECL(spx_word16_t *x); + VARDECL(spx_word16_t *y); + ALLOC(x, *in_len, spx_word16_t); + ALLOC(y, *out_len, spx_word16_t); */ + istride_save = st->in_stride; + ostride_save = st->out_stride; + for (i = 0; i < *in_len; i++) + x[i] = in[i * st->in_stride]; + st->in_stride = st->out_stride = 1; + speex_resampler_process_native (st, channel_index, x, in_len, y, out_len); + st->in_stride = istride_save; + st->out_stride = ostride_save; + for (i = 0; i < *out_len; i++) + out[i * st->out_stride] = WORD2INT (y[i]); +#else + spx_word16_t x[FIXED_STACK_ALLOC]; + spx_word16_t y[FIXED_STACK_ALLOC]; + spx_uint32_t ilen = *in_len, olen = *out_len; + + istride_save = st->in_stride; + ostride_save = st->out_stride; + while (ilen && olen) { + spx_uint32_t ichunk, ochunk; + + ichunk = ilen; + ochunk = olen; + if (ichunk > FIXED_STACK_ALLOC) + ichunk = FIXED_STACK_ALLOC; + if (ochunk > FIXED_STACK_ALLOC) + ochunk = FIXED_STACK_ALLOC; + for (i = 0; i < ichunk; i++) + x[i] = in[i * st->in_stride]; + st->in_stride = st->out_stride = 1; + speex_resampler_process_native (st, channel_index, x, &ichunk, y, &ochunk); + st->in_stride = istride_save; + st->out_stride = ostride_save; + for (i = 0; i < ochunk; i++) + out[i * st->out_stride] = WORD2INT (y[i]); + out += ochunk; + in += ichunk; + ilen -= ichunk; + olen -= ochunk; + } + *in_len -= ilen; + *out_len -= olen; +#endif + return RESAMPLER_ERR_SUCCESS; +} +#endif + +int +speex_resampler_process_interleaved_float (SpeexResamplerState * st, + const float *in, spx_uint32_t * in_len, float *out, spx_uint32_t * out_len) +{ + spx_uint32_t i; + int istride_save, ostride_save; + spx_uint32_t bak_len = *out_len; + + istride_save = st->in_stride; + ostride_save = st->out_stride; + st->in_stride = st->out_stride = st->nb_channels; + for (i = 0; i < st->nb_channels; i++) { + *out_len = bak_len; + speex_resampler_process_float (st, i, in + i, in_len, out + i, out_len); + } + st->in_stride = istride_save; + st->out_stride = ostride_save; + return RESAMPLER_ERR_SUCCESS; +} + + +int +speex_resampler_process_interleaved_int (SpeexResamplerState * st, + const spx_int16_t * in, spx_uint32_t * in_len, spx_int16_t * out, + spx_uint32_t * out_len) +{ + spx_uint32_t i; + int istride_save, ostride_save; + spx_uint32_t bak_len = *out_len; + + istride_save = st->in_stride; + ostride_save = st->out_stride; + st->in_stride = st->out_stride = st->nb_channels; + for (i = 0; i < st->nb_channels; i++) { + *out_len = bak_len; + speex_resampler_process_int (st, i, in + i, in_len, out + i, out_len); + } + st->in_stride = istride_save; + st->out_stride = ostride_save; + return RESAMPLER_ERR_SUCCESS; +} + +int +speex_resampler_set_rate (SpeexResamplerState * st, spx_uint32_t in_rate, + spx_uint32_t out_rate) +{ + return speex_resampler_set_rate_frac (st, in_rate, out_rate, in_rate, + out_rate); +} + +void +speex_resampler_get_rate (SpeexResamplerState * st, spx_uint32_t * in_rate, + spx_uint32_t * out_rate) +{ + *in_rate = st->in_rate; + *out_rate = st->out_rate; +} + +int +speex_resampler_set_rate_frac (SpeexResamplerState * st, spx_uint32_t ratio_num, + spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate) +{ + spx_uint32_t fact; + spx_uint32_t old_den; + spx_uint32_t i; + + if (st->in_rate == in_rate && st->out_rate == out_rate + && st->num_rate == ratio_num && st->den_rate == ratio_den) + return RESAMPLER_ERR_SUCCESS; + + old_den = st->den_rate; + st->in_rate = in_rate; + st->out_rate = out_rate; + st->num_rate = ratio_num; + st->den_rate = ratio_den; + /* FIXME: This is terribly inefficient, but who cares (at least for now)? */ + for (fact = 2; fact <= IMIN (st->num_rate, st->den_rate); fact++) { + while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) { + st->num_rate /= fact; + st->den_rate /= fact; + } + } + + if (old_den > 0) { + for (i = 0; i < st->nb_channels; i++) { + st->samp_frac_num[i] = st->samp_frac_num[i] * st->den_rate / old_den; + /* Safety net */ + if (st->samp_frac_num[i] >= st->den_rate) + st->samp_frac_num[i] = st->den_rate - 1; + } + } + + if (st->initialised) + update_filter (st); + return RESAMPLER_ERR_SUCCESS; +} + +void +speex_resampler_get_ratio (SpeexResamplerState * st, spx_uint32_t * ratio_num, + spx_uint32_t * ratio_den) +{ + *ratio_num = st->num_rate; + *ratio_den = st->den_rate; +} + +int +speex_resampler_set_quality (SpeexResamplerState * st, int quality) +{ + if (quality > 10 || quality < 0) + return RESAMPLER_ERR_INVALID_ARG; + if (st->quality == quality) + return RESAMPLER_ERR_SUCCESS; + st->quality = quality; + if (st->initialised) + update_filter (st); + return RESAMPLER_ERR_SUCCESS; +} + +void +speex_resampler_get_quality (SpeexResamplerState * st, int *quality) +{ + *quality = st->quality; +} + +void +speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride) +{ + st->in_stride = stride; +} + +void +speex_resampler_get_input_stride (SpeexResamplerState * st, + spx_uint32_t * stride) +{ + *stride = st->in_stride; +} + +void +speex_resampler_set_output_stride (SpeexResamplerState * st, + spx_uint32_t stride) +{ + st->out_stride = stride; +} + +void +speex_resampler_get_output_stride (SpeexResamplerState * st, + spx_uint32_t * stride) +{ + *stride = st->out_stride; +} + +int +speex_resampler_skip_zeros (SpeexResamplerState * st) +{ + spx_uint32_t i; + + for (i = 0; i < st->nb_channels; i++) + st->last_sample[i] = st->filt_len / 2; + return RESAMPLER_ERR_SUCCESS; +} + +int +speex_resampler_reset_mem (SpeexResamplerState * st) +{ + spx_uint32_t i; + + for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++) + st->mem[i] = 0; + return RESAMPLER_ERR_SUCCESS; +} + +const char * +speex_resampler_strerror (int err) +{ + switch (err) { + case RESAMPLER_ERR_SUCCESS: + return "Success."; + case RESAMPLER_ERR_ALLOC_FAILED: + return "Memory allocation failed."; + case RESAMPLER_ERR_BAD_STATE: + return "Bad resampler state."; + case RESAMPLER_ERR_INVALID_ARG: + return "Invalid argument."; + case RESAMPLER_ERR_PTR_OVERLAP: + return "Input and output buffers overlap."; + default: + return "Unknown error. Bad error code or strange version mismatch."; + } +} diff --git a/gst/speexresample/speex_resampler.h b/gst/speexresample/speex_resampler.h new file mode 100644 index 00000000..1dde54ac --- /dev/null +++ b/gst/speexresample/speex_resampler.h @@ -0,0 +1,325 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: speex_resampler.h + Resampling code + + The design goals of this code are: + - Very fast algorithm + - Low memory requirement + - Good *perceptual* quality (and not best SNR) + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + + +#ifndef SPEEX_RESAMPLER_H +#define SPEEX_RESAMPLER_H + +#ifdef OUTSIDE_SPEEX + +#include <glib.h> + +/********* WARNING: MENTAL SANITY ENDS HERE *************/ + +/* If the resampler is defined outside of Speex, we change the symbol names so that + there won't be any clash if linking with Speex later on. */ + +#define CAT_PREFIX2(a,b) a ## b +#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b) + +#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init) +#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac) +#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy) +#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float) +#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int) +#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float) +#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int) +#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate) +#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate) +#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac) +#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio) +#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality) +#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality) +#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride) +#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride) +#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride) +#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride) +#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros) +#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem) +#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror) + +#define spx_int16_t gint16 +#define spx_int32_t gint32 +#define spx_uint16_t guint16 +#define spx_uint32_t guint32 + +#else /* OUTSIDE_SPEEX */ + +#include "speex/speex_types.h" + +#endif /* OUTSIDE_SPEEX */ + +#ifdef __cplusplus +extern "C" { +#endif + +#define SPEEX_RESAMPLER_QUALITY_MAX 10 +#define SPEEX_RESAMPLER_QUALITY_MIN 0 +#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4 +#define SPEEX_RESAMPLER_QUALITY_VOIP 3 +#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5 + +enum { + RESAMPLER_ERR_SUCCESS = 0, + RESAMPLER_ERR_ALLOC_FAILED = 1, + RESAMPLER_ERR_BAD_STATE = 2, + RESAMPLER_ERR_INVALID_ARG = 3, + RESAMPLER_ERR_PTR_OVERLAP = 4, + + RESAMPLER_ERR_MAX_ERROR +}; + +struct SpeexResamplerState_; +typedef struct SpeexResamplerState_ SpeexResamplerState; + +/** Create a new resampler with integer input and output rates. + * @param nb_channels Number of channels to be processed + * @param in_rate Input sampling rate (integer number of Hz). + * @param out_rate Output sampling rate (integer number of Hz). + * @param quality Resampling quality between 0 and 10, where 0 has poor quality + * and 10 has very high quality. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, + spx_uint32_t in_rate, + spx_uint32_t out_rate, + int quality, + int *err); + +/** Create a new resampler with fractional input/output rates. The sampling + * rate ratio is an arbitrary rational number with both the numerator and + * denominator being 32-bit integers. + * @param nb_channels Number of channels to be processed + * @param ratio_num Numerator of the sampling rate ratio + * @param ratio_den Denominator of the sampling rate ratio + * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). + * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). + * @param quality Resampling quality between 0 and 10, where 0 has poor quality + * and 10 has very high quality. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate, + int quality, + int *err); + +/** Destroy a resampler state. + * @param st Resampler state + */ +void speex_resampler_destroy(SpeexResamplerState *st); + +/** Resample a float array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param channel_index Index of the channel to process for the multi-channel + * base (0 otherwise) + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the + * number of samples processed + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written + */ +int speex_resampler_process_float(SpeexResamplerState *st, + spx_uint32_t channel_index, + const float *in, + spx_uint32_t *in_len, + float *out, + spx_uint32_t *out_len); + +/** Resample an int array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param channel_index Index of the channel to process for the multi-channel + * base (0 otherwise) + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written + */ +int speex_resampler_process_int(SpeexResamplerState *st, + spx_uint32_t channel_index, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, + spx_uint32_t *out_len); + +/** Resample an interleaved float array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed. This is all per-channel. + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written. + * This is all per-channel. + */ +int speex_resampler_process_interleaved_float(SpeexResamplerState *st, + const float *in, + spx_uint32_t *in_len, + float *out, + spx_uint32_t *out_len); + +/** Resample an interleaved int array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed. This is all per-channel. + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written. + * This is all per-channel. + */ +int speex_resampler_process_interleaved_int(SpeexResamplerState *st, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, + spx_uint32_t *out_len); + +/** Set (change) the input/output sampling rates (integer value). + * @param st Resampler state + * @param in_rate Input sampling rate (integer number of Hz). + * @param out_rate Output sampling rate (integer number of Hz). + */ +int speex_resampler_set_rate(SpeexResamplerState *st, + spx_uint32_t in_rate, + spx_uint32_t out_rate); + +/** Get the current input/output sampling rates (integer value). + * @param st Resampler state + * @param in_rate Input sampling rate (integer number of Hz) copied. + * @param out_rate Output sampling rate (integer number of Hz) copied. + */ +void speex_resampler_get_rate(SpeexResamplerState *st, + spx_uint32_t *in_rate, + spx_uint32_t *out_rate); + +/** Set (change) the input/output sampling rates and resampling ratio + * (fractional values in Hz supported). + * @param st Resampler state + * @param ratio_num Numerator of the sampling rate ratio + * @param ratio_den Denominator of the sampling rate ratio + * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). + * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). + */ +int speex_resampler_set_rate_frac(SpeexResamplerState *st, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate); + +/** Get the current resampling ratio. This will be reduced to the least + * common denominator. + * @param st Resampler state + * @param ratio_num Numerator of the sampling rate ratio copied + * @param ratio_den Denominator of the sampling rate ratio copied + */ +void speex_resampler_get_ratio(SpeexResamplerState *st, + spx_uint32_t *ratio_num, + spx_uint32_t *ratio_den); + +/** Set (change) the conversion quality. + * @param st Resampler state + * @param quality Resampling quality between 0 and 10, where 0 has poor + * quality and 10 has very high quality. + */ +int speex_resampler_set_quality(SpeexResamplerState *st, + int quality); + +/** Get the conversion quality. + * @param st Resampler state + * @param quality Resampling quality between 0 and 10, where 0 has poor + * quality and 10 has very high quality. + */ +void speex_resampler_get_quality(SpeexResamplerState *st, + int *quality); + +/** Set (change) the input stride. + * @param st Resampler state + * @param stride Input stride + */ +void speex_resampler_set_input_stride(SpeexResamplerState *st, + spx_uint32_t stride); + +/** Get the input stride. + * @param st Resampler state + * @param stride Input stride copied + */ +void speex_resampler_get_input_stride(SpeexResamplerState *st, + spx_uint32_t *stride); + +/** Set (change) the output stride. + * @param st Resampler state + * @param stride Output stride + */ +void speex_resampler_set_output_stride(SpeexResamplerState *st, + spx_uint32_t stride); + +/** Get the output stride. + * @param st Resampler state copied + * @param stride Output stride + */ +void speex_resampler_get_output_stride(SpeexResamplerState *st, + spx_uint32_t *stride); + +/** Make sure that the first samples to go out of the resamplers don't have + * leading zeros. This is only useful before starting to use a newly created + * resampler. It is recommended to use that when resampling an audio file, as + * it will generate a file with the same length. For real-time processing, + * it is probably easier not to use this call (so that the output duration + * is the same for the first frame). + * @param st Resampler state + */ +int speex_resampler_skip_zeros(SpeexResamplerState *st); + +/** Reset a resampler so a new (unrelated) stream can be processed. + * @param st Resampler state + */ +int speex_resampler_reset_mem(SpeexResamplerState *st); + +/** Returns the English meaning for an error code + * @param err Error code + * @return English string + */ +const char *speex_resampler_strerror(int err); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/gst/speexresample/speex_resampler_float.c b/gst/speexresample/speex_resampler_float.c new file mode 100644 index 00000000..281e52d3 --- /dev/null +++ b/gst/speexresample/speex_resampler_float.c @@ -0,0 +1,24 @@ +/* GStreamer + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#define FLOATING_POINT +#define OUTSIDE_SPEEX +#define RANDOM_PREFIX resample_float + +#include "resample.c" diff --git a/gst/speexresample/speex_resampler_int.c b/gst/speexresample/speex_resampler_int.c new file mode 100644 index 00000000..c992f0a6 --- /dev/null +++ b/gst/speexresample/speex_resampler_int.c @@ -0,0 +1,24 @@ +/* GStreamer + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#define FIXED_POINT 1 +#define OUTSIDE_SPEEX 1 +#define RANDOM_PREFIX resample_int + +#include "resample.c" diff --git a/gst/speexresample/speex_resampler_wrapper.h b/gst/speexresample/speex_resampler_wrapper.h new file mode 100644 index 00000000..25f5576d --- /dev/null +++ b/gst/speexresample/speex_resampler_wrapper.h @@ -0,0 +1,80 @@ +/* GStreamer + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __SPEEX_RESAMPLER_WRAPPER_H__ +#define __SPEEX_RESAMPLER_WRAPPER_H__ + +#define SPEEX_RESAMPLER_QUALITY_MAX 10 +#define SPEEX_RESAMPLER_QUALITY_MIN 0 +#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4 +#define SPEEX_RESAMPLER_QUALITY_VOIP 3 +#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5 + +enum +{ + RESAMPLER_ERR_SUCCESS = 0, + RESAMPLER_ERR_ALLOC_FAILED = 1, + RESAMPLER_ERR_BAD_STATE = 2, + RESAMPLER_ERR_INVALID_ARG = 3, + RESAMPLER_ERR_PTR_OVERLAP = 4, + + RESAMPLER_ERR_MAX_ERROR +}; + +typedef struct SpeexResamplerState_ SpeexResamplerState; + +SpeexResamplerState *resample_float_resampler_init (guint32 nb_channels, + guint32 in_rate, guint32 out_rate, gint quality, gint * err); +SpeexResamplerState *resample_int_resampler_init (guint32 nb_channels, + guint32 in_rate, guint32 out_rate, gint quality, gint * err); + +#define resample_resampler_destroy resample_int_resampler_destroy +void resample_resampler_destroy (SpeexResamplerState * st); + +int resample_float_resampler_process_interleaved_float (SpeexResamplerState * + st, const gfloat * in, guint32 * in_len, gfloat * out, guint32 * out_len); +int resample_int_resampler_process_interleaved_int (SpeexResamplerState * st, + const gint16 * in, guint32 * in_len, gint16 * out, guint32 * out_len); + +int resample_float_resampler_set_rate (SpeexResamplerState * st, + guint32 in_rate, guint32 out_rate); +int resample_int_resampler_set_rate (SpeexResamplerState * st, + guint32 in_rate, guint32 out_rate); + +void resample_float_resampler_get_rate (SpeexResamplerState * st, + guint32 * in_rate, guint32 * out_rate); +void resample_int_resampler_get_rate (SpeexResamplerState * st, + guint32 * in_rate, guint32 * out_rate); + +void resample_float_resampler_get_ratio (SpeexResamplerState * st, + guint32 * ratio_num, guint32 * ratio_den); +void resample_int_resampler_get_ratio (SpeexResamplerState * st, + guint32 * ratio_num, guint32 * ratio_den); + +int resample_float_resampler_set_quality (SpeexResamplerState * st, + gint quality); +int resample_int_resampler_set_quality (SpeexResamplerState * st, gint quality); + +int resample_float_resampler_reset_mem (SpeexResamplerState * st); +int resample_int_resampler_reset_mem (SpeexResamplerState * st); + +#define resample_resampler_strerror resample_int_resampler_strerror +const char *resample_resampler_strerror (gint err); + +#endif /* __SPEEX_RESAMPLER_WRAPPER_H__ */ |