Bug 1341254 - Update libspeex_resampler to 79822c. r=karlt

MozReview-Commit-ID: EDYCyjrWmz1

--HG--
extra : rebase_source : 94d87c4b911ac646e755e03dc938da57ce237aad
This commit is contained in:
Paul Adenot 2017-03-06 17:16:23 +01:00
Родитель 6bb696b272
Коммит a01cea3a7b
13 изменённых файлов: 174 добавлений и 307 удалений

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@ -1,5 +1,5 @@
This source is from the Speex DSP library
(http://git.xiph.org/?p=speexdsp.git), from commit d60e75b2.
(http://git.xiph.org/?p=speexdsp.git), from commit 79822c.
It consists in the audio resampling code (resampler.c) and its header files
dependancies, imported into the tree using the update.sh script.

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@ -1,87 +0,0 @@
diff --git a/media/libspeex_resampler/fix-overflow.patch b/media/libspeex_resampler/fix-overflow.patch
new file mode 100644
index 0000000..e69de29
diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/src/resample.c
index a3859e3..d99595a 100644
--- a/media/libspeex_resampler/src/resample.c
+++ b/media/libspeex_resampler/src/resample.c
@@ -98,6 +98,10 @@ static void speex_free (void *ptr) {free(ptr);}
#define NULL 0
#endif
+#ifndef UINT32_MAX
+#define UINT32_MAX 4294967296U
+#endif
+
#include "simd_detect.h"
/* Numer of elements to allocate on the stack */
@@ -603,6 +607,22 @@ static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_in
return out_sample;
}
+static int _muldiv_safe(spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div)
+{
+ /* TODO: Could be simplified with 64 bits operation. */
+ spx_uint32_t major = value / div;
+ spx_uint32_t remainder = value % div;
+ return remainder <= UINT32_MAX / mul && major <= UINT32_MAX / mul &&
+ major * mul <= UINT32_MAX - remainder * mul / div;
+}
+
+static spx_uint32_t _muldiv(spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div)
+{
+ spx_uint32_t major = value / div;
+ spx_uint32_t remainder = value % div;
+ return remainder * mul / div + major * mul;
+}
+
static int update_filter(SpeexResamplerState *st)
{
spx_uint32_t old_length = st->filt_len;
@@ -620,8 +640,9 @@ static int update_filter(SpeexResamplerState *st)
{
/* 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;
+ if (!_muldiv_safe(st->filt_len,st->num_rate,st->den_rate))
+ goto fail;
+ st->filt_len = _muldiv(st->filt_len,st->num_rate,st->den_rate);
/* Round up to make sure we have a multiple of 8 for SSE */
st->filt_len = ((st->filt_len-1)&(~0x7))+8;
if (2*st->den_rate < st->num_rate)
@@ -1129,7 +1150,9 @@ EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t r
{
for (i=0;i<st->nb_channels;i++)
{
- st->samp_frac_num[i]=st->samp_frac_num[i]*st->den_rate/old_den;
+ if (!_muldiv_safe(st->samp_frac_num[i],st->den_rate,old_den))
+ return RESAMPLER_ERR_OVERFLOW;
+ st->samp_frac_num[i]= _muldiv(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;
diff --git a/media/libspeex_resampler/src/speex_resampler.h b/media/libspeex_resampler/src/speex_resampler.h
index 70abe52..1286872 100644
--- a/media/libspeex_resampler/src/speex_resampler.h
+++ b/media/libspeex_resampler/src/speex_resampler.h
@@ -106,7 +106,8 @@ enum {
RESAMPLER_ERR_BAD_STATE = 2,
RESAMPLER_ERR_INVALID_ARG = 3,
RESAMPLER_ERR_PTR_OVERLAP = 4,
-
+ RESAMPLER_ERR_OVERFLOW = 5,
+
RESAMPLER_ERR_MAX_ERROR
};
diff --git a/media/libspeex_resampler/update.sh b/media/libspeex_resampler/update.sh
index d4a025b..6950bc6 100644
--- a/media/libspeex_resampler/update.sh
+++ b/media/libspeex_resampler/update.sh
@@ -26,3 +26,4 @@ patch -p3 < set-skip-frac.patch
patch -p3 < hugemem.patch
patch -p3 < remove-empty-asm-clobber.patch
patch -p3 < handle-memory-error.patch
+patch -p3 < fix-overflow.patch

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@ -1,46 +0,0 @@
diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/src/resample.c
index 83ad119..a3859e3 100644
--- a/media/libspeex_resampler/src/resample.c
+++ b/media/libspeex_resampler/src/resample.c
@@ -811,6 +811,12 @@ EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
return NULL;
}
st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
+ if (!st)
+ {
+ if (err)
+ *err = RESAMPLER_ERR_ALLOC_FAILED;
+ return NULL;
+ }
st->initialised = 0;
st->started = 0;
st->in_rate = 0;
@@ -832,9 +838,12 @@ EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
st->buffer_size = 160;
/* Per channel data */
- st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t));
- st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
- st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
+ if (!(st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t))))
+ goto fail;
+ if (!(st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t))))
+ goto fail;
+ if (!(st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t))))
+ goto fail;
for (i=0;i<nb_channels;i++)
{
st->last_sample[i] = 0;
@@ -857,6 +866,12 @@ EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
*err = filter_err;
return st;
+
+fail:
+ if (err)
+ *err = RESAMPLER_ERR_ALLOC_FAILED;
+ speex_resampler_destroy(st);
+ return NULL;
}
EXPORT void speex_resampler_destroy(SpeexResamplerState *st)

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@ -2,7 +2,7 @@ diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/
--- a/media/libspeex_resampler/src/resample.c
+++ b/media/libspeex_resampler/src/resample.c
@@ -56,16 +56,18 @@
(e.g. 2/3), and get rid of the rounding operations in the inner loop.
(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.
*/
@ -20,7 +20,7 @@ diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/
#include "speex_resampler.h"
#include "arch.h"
#else /* OUTSIDE_SPEEX */
@@ -632,25 +634,26 @@ static int update_filter(SpeexResamplerS
@@ -643,25 +645,26 @@ static int update_filter(SpeexResamplerS
st->oversample >>= 1;
if (st->oversample < 1)
st->oversample = 1;
@ -28,7 +28,7 @@ diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/
/* up-sampling */
st->cutoff = quality_map[st->quality].upsample_bandwidth;
}
- /* Choose the resampling type that requires the least amount of memory */
-#ifdef RESAMPLE_FULL_SINC_TABLE
- use_direct = 1;
@ -54,3 +54,4 @@ diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/
goto fail;
min_sinc_table_length = st->filt_len*st->oversample+8;

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@ -15,7 +15,7 @@ diff --git a/media/libspeex_resampler/src/speex_resampler.h b/media/libspeex_res
/********* WARNING: MENTAL SANITY ENDS HERE *************/
/* If the resampler is defined outside of Speex, we change the symbol names so that
/* If the resampler is defined outside of Speex, we change the symbol names so that
there won't be any clash if linking with Speex later on. */
/* #define RANDOM_PREFIX your software name here */
@ -26,5 +26,5 @@ diff --git a/media/libspeex_resampler/src/speex_resampler.h b/media/libspeex_res
#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)

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@ -1,23 +1,25 @@
diff --git a/media/libspeex_resampler/src/resample.c b/media/libspeex_resampler/src/resample.c
--- a/media/libspeex_resampler/src/resample.c
+++ b/media/libspeex_resampler/src/resample.c
@@ -1146,17 +1146,19 @@ EXPORT int speex_resampler_set_rate_frac
}
}
@@ -1141,18 +1141,19 @@ EXPORT int speex_resampler_set_rate_frac
st->num_rate /= fact;
st->den_rate /= fact;
if (old_den > 0)
{
for (i=0;i<st->nb_channels;i++)
{
if (!_muldiv_safe(st->samp_frac_num[i],st->den_rate,old_den))
- if (_muldiv(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS)
- return RESAMPLER_ERR_OVERFLOW;
+ {
+ st->samp_frac_num[i] = st->den_rate-1;
+ if (_muldiv(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS) {
+ st->samp_frac_num[i] = st->den_rate-1;
+ }
st->samp_frac_num[i]= _muldiv(st->samp_frac_num[i],st->den_rate,old_den);
/* Safety net */
if (st->samp_frac_num[i] >= st->den_rate)
st->samp_frac_num[i] = st->den_rate-1;
}
}
if (st->initialised)
return update_filter(st);

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@ -7,18 +7,18 @@
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
@ -101,6 +101,8 @@ typedef spx_word32_t spx_sig_t;
#define SIG_SHIFT 14
#define GAIN_SHIFT 6
#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
#define VERY_SMALL 0
#define VERY_LARGE32 ((spx_word32_t)2147483647)
#define VERY_LARGE16 ((spx_word16_t)32767)
@ -203,18 +205,19 @@ typedef float spx_word32_t;
#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
#define WORD2INT(x) ((x) < -32767.5f ? -32768 : \
((x) > 32766.5f ? 32767 : (spx_int16_t)floor(.5 + (x))))
#endif
#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
/* 2 on TI C5x DSP */
#define BYTES_PER_CHAR 2
#define BYTES_PER_CHAR 2
#define BITS_PER_CHAR 16
#define LOG2_BITS_PER_CHAR 4
#else
#else
#define BYTES_PER_CHAR 1
#define BITS_PER_CHAR 8

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@ -7,18 +7,18 @@
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

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@ -1,6 +1,6 @@
/* Copyright (C) 2007-2008 Jean-Marc Valin
Copyright (C) 2008 Thorvald Natvig
File: resample.c
Arbitrary resampling code
@ -38,22 +38,22 @@
- Low memory requirement
- Good *perceptual* quality (and not best SNR)
Warning: This resampler is relatively new. Although I think I got rid of
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),
Center for Computer Research in Music and Acoustics (CCRMA),
Stanford University, 2007.
Web published at http://www-ccrma.stanford.edu/~jos/resample/.
Web published at http://ccrma.stanford.edu/~jos/resample/.
There is one main difference, though. This resampler uses cubic
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.
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.
*/
@ -85,12 +85,6 @@ static void speex_free (void *ptr) {free(ptr);}
#define M_PI 3.14159265358979323846
#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))
@ -118,7 +112,7 @@ struct SpeexResamplerState_ {
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;
@ -130,17 +124,17 @@ struct SpeexResamplerState_ {
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;
} ;
@ -182,7 +176,7 @@ static const double kaiser8_table[36] = {
0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
static const double kaiser6_table[36] = {
0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
@ -195,7 +189,7 @@ struct FuncDef {
const double *table;
int oversample;
};
static const struct FuncDef _KAISER12 = {kaiser12_table, 64};
#define KAISER12 (&_KAISER12)
/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
@ -217,7 +211,7 @@ struct QualityMapping {
/* This table maps conversion quality to internal parameters. There are two
reasons that explain why the up-sampling bandwidth is larger than the
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)
@ -243,7 +237,7 @@ static double compute_func(float x, const struct FuncDef *func)
{
float y, frac;
double interp[4];
int ind;
int ind;
y = x*func->oversample;
ind = (int)floor(y);
frac = (y-ind);
@ -254,7 +248,7 @@ static double compute_func(float x, const struct FuncDef *func)
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];
}
@ -493,7 +487,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3
sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
}
#endif
out[out_stride * out_sample++] = sum;
last_sample += int_advance;
samp_frac_num += frac_advance;
@ -559,7 +553,7 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3
sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
}
#endif
out[out_stride * out_sample++] = PSHR32(sum,15);
last_sample += int_advance;
samp_frac_num += frac_advance;
@ -607,20 +601,17 @@ static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_in
return out_sample;
}
static int _muldiv_safe(spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div)
static int _muldiv(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div)
{
/* TODO: Could be simplified with 64 bits operation. */
spx_uint32_t major = value / div;
spx_uint32_t remainder = value % div;
return remainder <= UINT32_MAX / mul && major <= UINT32_MAX / mul &&
major * mul <= UINT32_MAX - remainder * mul / div;
}
static spx_uint32_t _muldiv(spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div)
{
spx_uint32_t major = value / div;
spx_uint32_t remainder = value % div;
return remainder * mul / div + major * mul;
speex_assert(result);
spx_uint32_t major = value / div;
spx_uint32_t remainder = value % div;
/* TODO: Could use 64 bits operation to check for overflow. But only guaranteed in C99+ */
if (remainder > UINT32_MAX / mul || major > UINT32_MAX / mul
|| major * mul > UINT32_MAX - remainder * mul / div)
return RESAMPLER_ERR_OVERFLOW;
*result = remainder * mul / div + major * mul;
return RESAMPLER_ERR_SUCCESS;
}
static int update_filter(SpeexResamplerState *st)
@ -635,14 +626,13 @@ static int update_filter(SpeexResamplerState *st)
st->frac_advance = st->num_rate%st->den_rate;
st->oversample = quality_map[st->quality].oversample;
st->filt_len = quality_map[st->quality].base_length;
if (st->num_rate > st->den_rate)
{
/* down-sampling */
st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
if (!_muldiv_safe(st->filt_len,st->num_rate,st->den_rate))
if (_muldiv(&st->filt_len,st->filt_len,st->num_rate,st->den_rate) != RESAMPLER_ERR_SUCCESS)
goto fail;
st->filt_len = _muldiv(st->filt_len,st->num_rate,st->den_rate);
/* Round up to make sure we have a multiple of 8 for SSE */
st->filt_len = ((st->filt_len-1)&(~0x7))+8;
if (2*st->den_rate < st->num_rate)
@ -659,7 +649,7 @@ static int update_filter(SpeexResamplerState *st)
/* up-sampling */
st->cutoff = quality_map[st->quality].upsample_bandwidth;
}
use_direct =
#ifdef RESAMPLE_HUGEMEM
/* Choose the direct resampler, even with higher initialization costs,
@ -759,7 +749,7 @@ static int update_filter(SpeexResamplerState *st)
/*if (st->magic_samples[i])*/
{
/* Try and remove the magic samples as if nothing had happened */
/* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
olen = old_length + 2*st->magic_samples[i];
for (j=old_length-1+st->magic_samples[i];j--;)
@ -825,7 +815,7 @@ EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
SpeexResamplerState *st;
int filter_err;
if (quality > 10 || quality < 0)
if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || quality > 10 || quality < 0)
{
if (err)
*err = RESAMPLER_ERR_INVALID_ARG;
@ -850,14 +840,14 @@ EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
st->filt_len = 0;
st->mem = 0;
st->resampler_ptr = 0;
st->cutoff = 1.f;
st->nb_channels = nb_channels;
st->in_stride = 1;
st->out_stride = 1;
st->buffer_size = 160;
/* Per channel data */
if (!(st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t))))
goto fail;
@ -865,12 +855,6 @@ EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
goto fail;
if (!(st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t))))
goto fail;
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);
@ -912,17 +896,17 @@ static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t
int out_sample = 0;
spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
spx_uint32_t ilen;
st->started = 1;
/* Call the right resampler through the function ptr */
out_sample = st->resampler_ptr(st, channel_index, mem, in_len, out, out_len);
if (st->last_sample[channel_index] < (spx_int32_t)*in_len)
*in_len = st->last_sample[channel_index];
*out_len = out_sample;
st->last_sample[channel_index] -= *in_len;
ilen = *in_len;
for(j=0;j<N-1;++j)
@ -935,11 +919,11 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i
spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
const int N = st->filt_len;
speex_resampler_process_native(st, channel_index, &tmp_in_len, *out, &out_len);
st->magic_samples[channel_index] -= tmp_in_len;
/* If we couldn't process all "magic" input samples, save the rest for next time */
if (st->magic_samples[channel_index])
{
@ -965,13 +949,13 @@ EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t c
const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
const int istride = st->in_stride;
if (st->magic_samples[channel_index])
if (st->magic_samples[channel_index])
olen -= speex_resampler_magic(st, channel_index, &out, olen);
if (! st->magic_samples[channel_index]) {
while (ilen && olen) {
spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
spx_uint32_t ochunk = olen;
if (in) {
for(j=0;j<ichunk;++j)
x[j+filt_offs]=in[j*istride];
@ -1015,7 +999,7 @@ EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t cha
#endif
st->out_stride = 1;
while (ilen && olen) {
spx_word16_t *y = ystack;
spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
@ -1052,7 +1036,7 @@ EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t cha
#else
out[j*ostride_save] = WORD2INT(ystack[j]);
#endif
ilen -= ichunk;
olen -= ochunk;
out += (ochunk+omagic) * ostride_save;
@ -1088,7 +1072,7 @@ EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, co
st->out_stride = ostride_save;
return st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : RESAMPLER_ERR_SUCCESS;
}
EXPORT int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
{
spx_uint32_t i;
@ -1123,44 +1107,54 @@ EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_r
*out_rate = st->out_rate;
}
static inline spx_uint32_t _gcd(spx_uint32_t a, spx_uint32_t b)
{
while (b != 0)
{
spx_uint32_t temp = a;
a = b;
b = temp % b;
}
return a;
}
EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
{
spx_uint32_t fact;
spx_uint32_t old_den;
spx_uint32_t i;
if (ratio_num == 0 || ratio_den == 0)
return RESAMPLER_ERR_INVALID_ARG;
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;
}
}
fact = _gcd (st->num_rate, st->den_rate);
st->num_rate /= fact;
st->den_rate /= fact;
if (old_den > 0)
{
for (i=0;i<st->nb_channels;i++)
{
if (!_muldiv_safe(st->samp_frac_num[i],st->den_rate,old_den))
{
st->samp_frac_num[i] = st->den_rate-1;
if (_muldiv(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS) {
st->samp_frac_num[i] = st->den_rate-1;
}
st->samp_frac_num[i]= _muldiv(st->samp_frac_num[i],st->den_rate,old_den);
/* Safety net */
if (st->samp_frac_num[i] >= st->den_rate)
st->samp_frac_num[i] = st->den_rate-1;
}
}
if (st->initialised)
return update_filter(st);
return RESAMPLER_ERR_SUCCESS;

Просмотреть файл

@ -9,18 +9,18 @@
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

Просмотреть файл

@ -1,8 +1,8 @@
/* 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
@ -43,7 +43,7 @@
/********* WARNING: MENTAL SANITY ENDS HERE *************/
/* If the resampler is defined outside of Speex, we change the symbol names so that
/* If the resampler is defined outside of Speex, we change the symbol names so that
there won't be any clash if linking with Speex later on. */
/* #define RANDOM_PREFIX your software name here */
@ -54,7 +54,7 @@
#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)
@ -83,7 +83,9 @@
#define spx_int32_t int
#define spx_uint16_t unsigned short
#define spx_uint32_t unsigned int
#define speex_assert(cond)
#else /* OUTSIDE_SPEEX */
#include "speexdsp_types.h"
@ -123,14 +125,14 @@ typedef struct SpeexResamplerState_ SpeexResamplerState;
* @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,
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
/** 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
@ -142,11 +144,11 @@ SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
* @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,
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);
@ -157,24 +159,24 @@ 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
* @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
* @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,
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
* @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
@ -182,11 +184,11 @@ int speex_resampler_process_float(SpeexResamplerState *st,
* @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,
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.
@ -198,10 +200,10 @@ int speex_resampler_process_int(SpeexResamplerState *st,
* @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,
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.
@ -213,10 +215,10 @@ int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
* @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,
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).
@ -224,8 +226,8 @@ int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
* @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,
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).
@ -233,11 +235,11 @@ int speex_resampler_set_rate(SpeexResamplerState *st,
* @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,
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
/** 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
@ -245,10 +247,10 @@ void speex_resampler_get_rate(SpeexResamplerState *st,
* @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,
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
@ -257,52 +259,52 @@ int speex_resampler_set_rate_frac(SpeexResamplerState *st,
* @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,
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
* @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 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
* @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,
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,
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,
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,
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,
void speex_resampler_get_output_stride(SpeexResamplerState *st,
spx_uint32_t *stride);
/** Get the latency introduced by the resampler measured in input samples.
@ -315,8 +317,8 @@ int speex_resampler_get_input_latency(SpeexResamplerState *st);
*/
int speex_resampler_get_output_latency(SpeexResamplerState *st);
/** Make sure that the first samples to go out of the resamplers don't have
* leading zeros. This is only useful before starting to use a newly created
/** 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

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@ -7,18 +7,18 @@
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
@ -101,7 +101,7 @@
#endif
#if defined(VAR_ARRAYS)
#define VARDECL(var)
#define VARDECL(var)
#define ALLOC(var, size, type) type var[size]
#elif defined(USE_ALLOCA)
#define VARDECL(var) var

Просмотреть файл

@ -25,6 +25,4 @@ patch -p3 < simd-detect-runtime.patch
patch -p3 < set-skip-frac.patch
patch -p3 < hugemem.patch
patch -p3 < remove-empty-asm-clobber.patch
patch -p3 < handle-memory-error.patch
patch -p3 < fix-overflow.patch
patch -p3 < set-rate-overflow-no-return.patch