[asterisk-commits] dvossel: branch dvossel/hd_conferencing_ftw r308197 - in /team/dvossel/hd_con...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Feb 16 16:58:19 CST 2011
Author: dvossel
Date: Wed Feb 16 16:58:16 2011
New Revision: 308197
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=308197
Log:
Remove codec_resample dependency by using SPEEX slin resampling
Added:
team/dvossel/hd_conferencing_ftw/codecs/speex/
team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h (with props)
team/dvossel/hd_conferencing_ftw/codecs/speex/fixed_generic.h (with props)
team/dvossel/hd_conferencing_ftw/codecs/speex/resample.c (with props)
team/dvossel/hd_conferencing_ftw/codecs/speex/resample_sse.h (with props)
team/dvossel/hd_conferencing_ftw/codecs/speex/speex_resampler.h (with props)
team/dvossel/hd_conferencing_ftw/codecs/speex/stack_alloc.h (with props)
Modified:
team/dvossel/hd_conferencing_ftw/codecs/Makefile
team/dvossel/hd_conferencing_ftw/codecs/codec_resample.c
Modified: team/dvossel/hd_conferencing_ftw/codecs/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/codecs/Makefile?view=diff&rev=308197&r1=308196&r2=308197
==============================================================================
--- team/dvossel/hd_conferencing_ftw/codecs/Makefile (original)
+++ team/dvossel/hd_conferencing_ftw/codecs/Makefile Wed Feb 16 16:58:16 2011
@@ -29,11 +29,13 @@
$(if $(filter codec_gsm,$(EMBEDDED_MODS)),modules.link,codec_gsm.so): gsm/lib/libgsm.a
endif
+
clean::
$(MAKE) -C gsm clean
$(MAKE) -C lpc10 clean
$(MAKE) -C ilbc clean
rm -f g722/*.[oa]
+ rm -f speex/*.[oa]
gsm/lib/libgsm.a:
@mkdir -p gsm/lib
@@ -47,7 +49,17 @@
$(LIBILBC):
@$(MAKE) -C ilbc all _ASTCFLAGS="$(filter-out -Wmissing-prototypes -Wmissing-declarations -Wshadow,$(_ASTCFLAGS)) $(AST_NO_STRICT_OVERFLOW)"
+
$(if $(filter codec_ilbc,$(EMBEDDED_MODS)),modules.link,codec_ilbc.so): $(LIBILBC)
$(if $(filter codec_g722,$(EMBEDDED_MODS)),modules.link,codec_g722.so): g722/g722_encode.o g722/g722_decode.o
g722/g722_encode.o g722/g722_decode.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,codec_g722)
+
+ifeq ($(BUILD_CPU),x86_64)
+SPEEX_RESAMPLE_CFLAGS:=-fPIC
+else
+SPEEX_RESAMPLE_CFLAGS:=
+endif
+
+$(if $(filter codec_resample,$(EMBEDDED_MODS)),modules.link,codec_resample.so): speex/resample.o
+speex/resample.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,codec_resample) $(SPEEX_RESAMPLE_CFLAGS)
Modified: team/dvossel/hd_conferencing_ftw/codecs/codec_resample.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/codecs/codec_resample.c?view=diff&rev=308197&r1=308196&r2=308197
==============================================================================
--- team/dvossel/hd_conferencing_ftw/codecs/codec_resample.c (original)
+++ team/dvossel/hd_conferencing_ftw/codecs/codec_resample.c Wed Feb 16 16:58:16 2011
@@ -1,9 +1,10 @@
/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 2007, Digium, Inc.
+ * Copyright (C) 2011, Digium, Inc.
*
* Russell Bryant <russell at digium.com>
+ * David Vossel <dvossel at digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
@@ -21,9 +22,6 @@
*
* \brief Resample slinear audio
*
- * \note To install libresample, check it out of the following repository:
- * <code>$ svn co http://svn.digium.com/svn/thirdparty/libresample/trunk</code>
- *
* \ingroup codecs
*/
@@ -32,26 +30,13 @@
***/
#include "asterisk.h"
+#include "speex/speex_resampler.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-/* These are for SHRT_MAX and FLT_MAX -- { */
-#if defined(__Darwin__) || defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__CYGWIN__)
-#include <float.h>
-#else
-#include <values.h>
-#endif
-#include <limits.h>
-/* } */
-
-#include <libresample.h>
-
#include "asterisk/module.h"
#include "asterisk/translate.h"
-
#include "asterisk/slin.h"
-
-#define RESAMPLER_QUALITY 1
#define OUTBUF_SIZE 8096
@@ -70,18 +55,15 @@
};
struct resamp_pvt {
- void *resampler;
- float resample_factor;
+ SpeexResamplerState *resampler;
};
static int resamp_new(struct ast_trans_pvt *pvt)
{
struct resamp_pvt *resamp_pvt = pvt->pvt;
+ int err;
- resamp_pvt->resample_factor = (float) ast_format_rate(&pvt->t->dst_format) / ast_format_rate(&pvt->t->src_format);
-
- if (!(resamp_pvt->resampler = resample_open(RESAMPLER_QUALITY, resamp_pvt->resample_factor, resamp_pvt->resample_factor))) {
- ast_log(LOG_WARNING, "Failed to create resample translator from %d to %d resamp factor %f\n", ast_format_rate(&pvt->t->dst_format), ast_format_rate(&pvt->t->src_format), resamp_pvt->resample_factor);
+ if (!(resamp_pvt->resampler = speex_resampler_init(1, ast_format_rate(&pvt->t->src_format), ast_format_rate(&pvt->t->dst_format), 5, &err))) {
return -1;
}
@@ -91,63 +73,27 @@
static void resamp_destroy(struct ast_trans_pvt *pvt)
{
struct resamp_pvt *resamp_pvt = pvt->pvt;
+ speex_resampler_destroy(resamp_pvt->resampler);
- if (resamp_pvt->resampler)
- resample_close(resamp_pvt->resampler);
-}
-
-static int resample_frame(struct ast_trans_pvt *pvt,
- void *resampler, float resample_factor, struct ast_frame *f)
-{
- int total_in_buf_used = 0;
- int total_out_buf_used = 0;
- int16_t *in_buf = (int16_t *) f->data.ptr;
- int16_t *out_buf = pvt->outbuf.i16 + pvt->samples;
- float in_buf_f[f->samples];
- float out_buf_f[2048];
- int res = 0;
- int i;
-
- for (i = 0; i < f->samples; i++)
- in_buf_f[i] = in_buf[i] * (FLT_MAX / SHRT_MAX);
-
- while (total_in_buf_used < f->samples) {
- int in_buf_used, out_buf_used;
-
- out_buf_used = resample_process(resampler, resample_factor,
- &in_buf_f[total_in_buf_used], f->samples - total_in_buf_used,
- 0, &in_buf_used,
- &out_buf_f[total_out_buf_used], ARRAY_LEN(out_buf_f) - total_out_buf_used);
-
- if (out_buf_used < 0)
- break;
-
- total_out_buf_used += out_buf_used;
- total_in_buf_used += in_buf_used;
-
- if (total_out_buf_used == ARRAY_LEN(out_buf_f)) {
- ast_log(LOG_ERROR, "Output buffer filled ... need to increase its size\n");
- res = -1;
- break;
- }
- }
-
- for (i = 0; i < total_out_buf_used; i++)
- out_buf[i] = out_buf_f[i] * (SHRT_MAX / FLT_MAX);
-
- pvt->samples += total_out_buf_used;
- pvt->datalen += (total_out_buf_used * sizeof(int16_t));
-
- return res;
}
static int resamp_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
struct resamp_pvt *resamp_pvt = pvt->pvt;
- void *resampler = resamp_pvt->resampler;
- float resample_factor = resamp_pvt->resample_factor;
+ unsigned int out_samples = (OUTBUF_SIZE / sizeof(int16_t)) - pvt->samples;
+ unsigned int in_samples = f->samples;
- return resample_frame(pvt, resampler, resample_factor, f);
+ speex_resampler_process_int(resamp_pvt->resampler,
+ 0,
+ f->data.ptr,
+ &in_samples,
+ pvt->outbuf.i16 + pvt->samples,
+ &out_samples);
+
+ pvt->samples += out_samples;
+ pvt->datalen += out_samples * 2;
+
+ return 0;
}
static int unload_module(void)
@@ -155,7 +101,7 @@
int res = 0;
int idx;
- for (idx = 0; idx < ARRAY_LEN(translators); idx++) {
+ for (idx = 0; idx < trans_size; idx++) {
res |= ast_unregister_translator(&translators[idx]);
}
ast_free(translators);
Added: team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h?view=auto&rev=308197
==============================================================================
--- team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h (added)
+++ team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h Wed Feb 16 16:58:16 2011
@@ -1,0 +1,241 @@
+/* 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
+
+#define FIXED_POINT
+
+/* 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
+
+#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */
+#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 16-bit value. */
+#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 16-bit value. */
+#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */
+#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 32-bit value. */
+#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 32-bit value. */
+#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */
+
+#ifdef FIXED_POINT
+
+typedef spx_int16_t spx_word16_t;
+typedef spx_int32_t spx_word32_t;
+typedef spx_word32_t spx_mem_t;
+typedef spx_word16_t spx_coef_t;
+typedef spx_word16_t spx_lsp_t;
+typedef spx_word32_t spx_sig_t;
+
+#define Q15ONE 32767
+
+#define LPC_SCALING 8192
+#define SIG_SCALING 16384
+#define LSP_SCALING 8192.
+#define GAMMA_SCALING 32768.
+#define GAIN_SCALING 64
+#define GAIN_SCALING_1 0.015625
+
+#define LPC_SHIFT 13
+#define LSP_SHIFT 13
+#define SIG_SHIFT 14
+#define GAIN_SHIFT 6
+
+#define VERY_SMALL 0
+#define VERY_LARGE32 ((spx_word32_t)2147483647)
+#define VERY_LARGE16 ((spx_word16_t)32767)
+#define Q15_ONE ((spx_word16_t)32767)
+
+
+#ifdef FIXED_DEBUG
+#include "fixed_debug.h"
+#else
+
+#include "fixed_generic.h"
+
+#ifdef ARM5E_ASM
+#include "fixed_arm5e.h"
+#elif defined (ARM4_ASM)
+#include "fixed_arm4.h"
+#elif defined (BFIN_ASM)
+#include "fixed_bfin.h"
+#endif
+
+#endif
+
+
+#else
+
+typedef float spx_mem_t;
+typedef float spx_coef_t;
+typedef float spx_lsp_t;
+typedef float spx_sig_t;
+typedef float spx_word16_t;
+typedef float spx_word32_t;
+
+#define Q15ONE 1.0f
+#define LPC_SCALING 1.f
+#define SIG_SCALING 1.f
+#define LSP_SCALING 1.f
+#define GAMMA_SCALING 1.f
+#define GAIN_SCALING 1.f
+#define GAIN_SCALING_1 1.f
+
+
+#define VERY_SMALL 1e-15f
+#define VERY_LARGE32 1e15f
+#define VERY_LARGE16 1e15f
+#define Q15_ONE ((spx_word16_t)1.f)
+
+#define QCONST16(x,bits) (x)
+#define QCONST32(x,bits) (x)
+
+#define NEG16(x) (-(x))
+#define NEG32(x) (-(x))
+#define EXTRACT16(x) (x)
+#define EXTEND32(x) (x)
+#define SHR16(a,shift) (a)
+#define SHL16(a,shift) (a)
+#define SHR32(a,shift) (a)
+#define SHL32(a,shift) (a)
+#define PSHR16(a,shift) (a)
+#define PSHR32(a,shift) (a)
+#define VSHR32(a,shift) (a)
+#define SATURATE16(x,a) (x)
+#define SATURATE32(x,a) (x)
+
+#define 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
+extern long long spx_mips;
+#endif
+
+
+#endif
Propchange: team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/dvossel/hd_conferencing_ftw/codecs/speex/arch.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: team/dvossel/hd_conferencing_ftw/codecs/speex/fixed_generic.h
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/codecs/speex/fixed_generic.h?view=auto&rev=308197
==============================================================================
--- team/dvossel/hd_conferencing_ftw/codecs/speex/fixed_generic.h (added)
+++ team/dvossel/hd_conferencing_ftw/codecs/speex/fixed_generic.h Wed Feb 16 16:58:16 2011
@@ -1,0 +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)+((EXTEND32(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)+((EXTEND32(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
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Added: team/dvossel/hd_conferencing_ftw/codecs/speex/resample.c
URL: http://svnview.digium.com/svn/asterisk/team/dvossel/hd_conferencing_ftw/codecs/speex/resample.c?view=auto&rev=308197
==============================================================================
--- team/dvossel/hd_conferencing_ftw/codecs/speex/resample.c (added)
+++ team/dvossel/hd_conferencing_ftw/codecs/speex/resample.c Wed Feb 16 16:58:16 2011
@@ -1,0 +1,1124 @@
+/* Copyright (C) 2007-2008 Jean-Marc Valin
+ Copyright (C) 2008 Thorvald Natvig
+
+ File: resample.c
+ Arbitrary resampling code
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions are
+ met:
+
+ 1. Redistributions of source code must retain the above copyright notice,
+ this list of conditions and the following disclaimer.
+
+ 2. Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+
+ 3. The name of the author may not be used to endorse or promote products
+ derived from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
+ IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
+ OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
+ INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+ ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/*
+ The design goals of this code are:
+ - Very fast algorithm
+ - SIMD-friendly algorithm
+ - Low memory requirement
+ - Good *perceptual* quality (and not best SNR)
+
+ Warning: This resampler is relatively new. Although I think I got rid of
+ all the major bugs and I don't expect the API to change anymore, there
+ may be something I've missed. So use with caution.
+
+ This algorithm is based on this original resampling algorithm:
+ Smith, Julius O. Digital Audio Resampling Home Page
+ Center for Computer Research in Music and Acoustics (CCRMA),
+ Stanford University, 2007.
+ Web published at http://www-ccrma.stanford.edu/~jos/resample/.
+
+ There is one main difference, though. This resampler uses cubic
+ interpolation instead of linear interpolation in the above paper. This
+ makes the table much smaller and makes it possible to compute that table
+ on a per-stream basis. In turn, being able to tweak the table for each
+ stream makes it possible to both reduce complexity on simple ratios
+ (e.g. 2/3), and get rid of the rounding operations in the inner loop.
+ The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#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"
+
+#include "stack_alloc.h"
+#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 IMAX(a,b) ((a) > (b) ? (a) : (b))
+#define IMIN(a,b) ((a) < (b) ? (a) : (b))
+
+#ifndef NULL
+#define NULL 0
+#endif
+
+#ifdef _USE_SSE
+#include "resample_sse.h"
+#endif
+
+/* Numer of elements to allocate on the stack */
+#ifdef VAR_ARRAYS
+#define FIXED_STACK_ALLOC 8192
+#else
+#define FIXED_STACK_ALLOC 1024
+#endif
+
+typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
+
+struct SpeexResamplerState_ {
+ spx_uint32_t in_rate;
+ spx_uint32_t out_rate;
+ spx_uint32_t num_rate;
+ spx_uint32_t den_rate;
+
+ int quality;
+ spx_uint32_t nb_channels;
+ spx_uint32_t filt_len;
+ spx_uint32_t mem_alloc_size;
+ spx_uint32_t buffer_size;
+ int int_advance;
+ int frac_advance;
+ float cutoff;
+ spx_uint32_t oversample;
+ int initialised;
+ int started;
+
+ /* These are per-channel */
+ spx_int32_t *last_sample;
+ spx_uint32_t *samp_frac_num;
+ spx_uint32_t *magic_samples;
+
+ spx_word16_t *mem;
+ spx_word16_t *sinc_table;
+ spx_uint32_t sinc_table_length;
+ resampler_basic_func resampler_ptr;
+
+ int in_stride;
+ int out_stride;
+} ;
+
+static 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)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const spx_word16_t *sinc_table = st->sinc_table;
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ spx_word32_t sum;
+ int j;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
+ const spx_word16_t *iptr = & in[last_sample];
+
+#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
+ float accum[4] = {0,0,0,0};
+
+ for(j=0;j<N;j+=4) {
+ accum[0] += sinc[j]*iptr[j];
+ accum[1] += sinc[j+1]*iptr[j+1];
+ accum[2] += sinc[j+2]*iptr[j+2];
+ accum[3] += sinc[j+3]*iptr[j+3];
+ }
+ sum = accum[0] + accum[1] + accum[2] + accum[3];
+#else
+ sum = inner_product_single(sinc, iptr, N);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32(sum, 15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const spx_word16_t *sinc_table = st->sinc_table;
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ double sum;
+ int j;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
+ const spx_word16_t *iptr = & in[last_sample];
+
+#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
+ double accum[4] = {0,0,0,0};
+
+ for(j=0;j<N;j+=4) {
+ accum[0] += sinc[j]*iptr[j];
+ accum[1] += sinc[j+1]*iptr[j+1];
+ accum[2] += sinc[j+2]*iptr[j+2];
+ accum[3] += sinc[j+3]*iptr[j+3];
+ }
+ sum = accum[0] + accum[1] + accum[2] + accum[3];
+#else
+ sum = inner_product_double(sinc, iptr, N);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32(sum, 15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate)
+ {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+#endif
+
+static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ int j;
+ spx_word32_t sum;
+
+ while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
+ {
+ const spx_word16_t *iptr = & in[last_sample];
+
+ const int offset = samp_frac_num*st->oversample/st->den_rate;
+#ifdef FIXED_POINT
+ const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
+#else
+ const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
+#endif
+ spx_word16_t interp[4];
+
+
+#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
+ spx_word32_t accum[4] = {0,0,0,0};
+
[... 1330 lines stripped ...]
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