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lavr: resampling: add support for s32p, fltp, and dblp internal sample formats 

Based partially on implementation by Michael Niedermayer <michaelni@gmx.at> in
libswresample in FFmpeg. See commits:
7f1ae79d38
24ab1abfb6
tags/n1.0
Justin Ruggles 13 years ago
parent
372647aed0
commit
6410397600
3 changed files with 199 additions and 88 deletions
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  1. +76
    -87
      libavresample/resample.c
  2. +102
    -0
      libavresample/resample_template.c
  3. +21
    -1
      libavresample/utils.c

+ 76
- 87
libavresample/resample.c View File

@@ -24,34 +24,10 @@
#include "internal.h"
#include "audio_data.h"

#ifdef CONFIG_RESAMPLE_FLT
/* float template */
#define FILTER_SHIFT 0
#define FELEM float
#define FELEM2 float
#define FELEML float
#elifdef CONFIG_RESAMPLE_S32
/* s32 template */
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#else
/* s16 template */
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#endif

struct ResampleContext {
AVAudioResampleContext *avr;
AudioData *buffer;
FELEM *filter_bank;
uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
@@ -65,8 +41,32 @@ struct ResampleContext {
enum AVResampleFilterType filter_type;
int kaiser_beta;
double factor;
void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
int dst_index, const void *src0, int src_size,
int index, int frac);
};


/* double template */
#define CONFIG_RESAMPLE_DBL
#include "resample_template.c"
#undef CONFIG_RESAMPLE_DBL

/* float template */
#define CONFIG_RESAMPLE_FLT
#include "resample_template.c"
#undef CONFIG_RESAMPLE_FLT

/* s32 template */
#define CONFIG_RESAMPLE_S32
#include "resample_template.c"
#undef CONFIG_RESAMPLE_S32

/* s16 template */
#include "resample_template.c"


/**
* 0th order modified bessel function of the first kind.
*/
@@ -98,13 +98,13 @@ static double bessel(double x)
* @param kaiser_beta kaiser window beta
* @return 0 on success, negative AVERROR code on failure
*/
static int build_filter(FELEM *filter, double factor, int tap_count,
int phase_count, int scale, int filter_type,
int kaiser_beta)
static int build_filter(ResampleContext *c)
{
int ph, i;
double x, y, w;
double x, y, w, factor;
double *tab;
int tap_count = c->filter_length;
int phase_count = 1 << c->phase_shift;
const int center = (tap_count - 1) / 2;

tab = av_malloc(tap_count * sizeof(*tab));
@@ -112,8 +112,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
return AVERROR(ENOMEM);

/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
factor = FFMIN(c->factor, 1.0);

for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
@@ -121,7 +120,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch (filter_type) {
switch (c->filter_type) {
case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
const float d = -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
@@ -137,23 +136,18 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
break;
case AV_RESAMPLE_FILTER_TYPE_KAISER:
w = 2.0 * x / (factor * tap_count * M_PI);
y *= bessel(kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
break;
}

tab[i] = y;
norm += y;
}

/* normalize so that an uniform color remains the same */
for (i = 0; i < tap_count; i++) {
#ifdef CONFIG_RESAMPLE_FLT
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
FELEM_MIN, FELEM_MAX);
#endif
}
for (i = 0; i < tap_count; i++)
tab[i] = tab[i] / norm;

c->set_filter(c->filter_bank, tab, ph, tap_count);
}

av_free(tab);
@@ -167,9 +161,12 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
int in_rate = avr->in_sample_rate;
double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
int phase_count = 1 << avr->phase_shift;
int felem_size;

/* TODO: add support for s32 and float internal formats */
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"resampling: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
@@ -188,17 +185,37 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
c->filter_type = avr->filter_type;
c->kaiser_beta = avr->kaiser_beta;

c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
switch (avr->internal_sample_fmt) {
case AV_SAMPLE_FMT_DBLP:
c->resample_one = resample_one_dbl;
c->set_filter = set_filter_dbl;
break;
case AV_SAMPLE_FMT_FLTP:
c->resample_one = resample_one_flt;
c->set_filter = set_filter_flt;
break;
case AV_SAMPLE_FMT_S32P:
c->resample_one = resample_one_s32;
c->set_filter = set_filter_s32;
break;
case AV_SAMPLE_FMT_S16P:
c->resample_one = resample_one_s16;
c->set_filter = set_filter_s16;
break;
}

felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
if (!c->filter_bank)
goto error;

if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
1 << FILTER_SHIFT, c->filter_type, c->kaiser_beta) < 0)
if (build_filter(c) < 0)
goto error;

memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
c->filter_bank, (c->filter_length - 1) * felem_size);
memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
&c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);

c->compensation_distance = 0;
if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
@@ -312,10 +329,10 @@ reinit_fail:
return ret;
}

static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
static int resample(ResampleContext *c, void *dst, const void *src,
int *consumed, int src_size, int dst_size, int update_ctx)
{
int dst_index, i;
int dst_index;
int index = c->index;
int frac = c->frac;
int dst_incr_frac = c->dst_incr % c->src_incr;
@@ -335,7 +352,7 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,

if (dst) {
for(dst_index = 0; dst_index < dst_size; dst_index++) {
dst[dst_index] = src[index2 >> 32];
c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
index2 += incr;
}
} else {
@@ -346,42 +363,14 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
} else {
for (dst_index = 0; dst_index < dst_size; dst_index++) {
FELEM *filter = c->filter_bank +
c->filter_length * (index & c->phase_mask);
int sample_index = index >> c->phase_shift;

if (!dst && (sample_index + c->filter_length > src_size ||
-sample_index >= src_size))
if (sample_index + c->filter_length > src_size ||
-sample_index >= src_size)
break;

if (dst) {
FELEM2 val = 0;

if (sample_index < 0) {
for (i = 0; i < c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] *
(FELEM2)filter[i];
} else if (sample_index + c->filter_length > src_size) {
break;
} else if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[abs(sample_index + i)] * (FELEM2)filter[i];
v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
}
val += (v2 - val) * (FELEML)frac / c->src_incr;
} else {
for (i = 0; i < c->filter_length; i++)
val += src[sample_index + i] * (FELEM2)filter[i];
}

#ifdef CONFIG_RESAMPLE_FLT
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = av_clip_int16(val);
#endif
}
if (dst)
c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);

frac += dst_incr_frac;
index += dst_incr;
@@ -452,8 +441,8 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,

/* resample each channel plane */
for (ch = 0; ch < c->buffer->channels; ch++) {
out_samples = resample(c, (int16_t *)dst->data[ch],
(const int16_t *)c->buffer->data[ch], consumed,
out_samples = resample(c, (void *)dst->data[ch],
(const void *)c->buffer->data[ch], consumed,
c->buffer->nb_samples, dst->allocated_samples,
ch + 1 == c->buffer->channels);
}


+ 102
- 0
libavresample/resample_template.c View File

@@ -0,0 +1,102 @@
/*
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#if defined(CONFIG_RESAMPLE_DBL)
#define SET_TYPE(func) func ## _dbl
#define FELEM double
#define FELEM2 double
#define FELEML double
#define OUT(d, v) d = v
#define DBL_TO_FELEM(d, v) d = v
#elif defined(CONFIG_RESAMPLE_FLT)
#define SET_TYPE(func) func ## _flt
#define FELEM float
#define FELEM2 float
#define FELEML float
#define OUT(d, v) d = v
#define DBL_TO_FELEM(d, v) d = v
#elif defined(CONFIG_RESAMPLE_S32)
#define SET_TYPE(func) func ## _s32
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
#else
#define SET_TYPE(func) func ## _s16
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
#endif

static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter,
void *dst0, int dst_index, const void *src0,
int src_size, int index, int frac)
{
FELEM *dst = dst0;
const FELEM *src = src0;

if (no_filter) {
dst[dst_index] = src[index];
} else {
int i;
int sample_index = index >> c->phase_shift;
FELEM2 val = 0;
FELEM *filter = ((FELEM *)c->filter_bank) +
c->filter_length * (index & c->phase_mask);

if (sample_index < 0) {
for (i = 0; i < c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] *
(FELEM2)filter[i];
} else if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[abs(sample_index + i)] * (FELEM2)filter[i];
v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
}
val += (v2 - val) * (FELEML)frac / c->src_incr;
} else {
for (i = 0; i < c->filter_length; i++)
val += src[sample_index + i] * (FELEM2)filter[i];
}

OUT(dst[dst_index], val);
}
}

static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
int tap_count)
{
int i;
FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
for (i = 0; i < tap_count; i++) {
DBL_TO_FELEM(filter[i], tab[i]);
}
}

#undef SET_TYPE
#undef FELEM
#undef FELEM2
#undef FELEML
#undef OUT
#undef DBL_TO_FELEM

+ 21
- 1
libavresample/utils.c View File

@@ -64,10 +64,30 @@ int avresample_open(AVAudioResampleContext *avr)
enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
av_get_bytes_per_sample(out_fmt));
if (avr->resample_needed || max_bps <= 2) {
if (max_bps <= 2) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
} else {
if (max_bps <= 4) {
if (in_fmt == AV_SAMPLE_FMT_S32P ||
out_fmt == AV_SAMPLE_FMT_S32P) {
if (in_fmt == AV_SAMPLE_FMT_FLTP ||
out_fmt == AV_SAMPLE_FMT_FLTP) {
/* if one is s32 and the other is flt, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
} else {
/* if one is s32 and the other is s32, s16, or u8, use s32 */
avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
}
} else {
/* if one is flt and the other is flt, s16 or u8, use flt */
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
} else {
/* if either is dbl, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
}
}
av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));


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