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swresample: add exact_rational option

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give high quality resampling
as good as with linear_interp=on
as fast as without linear_interp=on
tested visually with ffplay
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000, showcqt=gamma=5"
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000:linear_interp=on, showcqt=gamma=5"
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000:exact_rational=on, showcqt=gamma=5"

slightly speed improvement
for fair comparison with -cpuflags 0
audio.wav is ~ 1 hour 44100 stereo 16bit wav file
ffmpeg -i audio.wav -af aresample=osr=48000 -f null -
        old         new
real    13.498s     13.121s
user    13.364s     12.987s
sys      0.131s      0.129s

linear_interp=on
        old         new
real    23.035s     23.050s
user    22.907s     22.917s
sys      0.119s     0.125s

exact_rational=on
real    12.418s
user    12.298s
sys      0.114s

possibility to decrease memory usage if soft compensation is ignored

Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
This commit is contained in:
Muhammad Faiz
2016-06-12 05:19:20 +07:00
parent 5ca44ebd99
commit b8c6e5a661
10 changed files with 78 additions and 30 deletions
Download Patch File Download Diff File Expand all files Collapse all files

15
libswresample/arm/resample_init.c
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@@ -44,11 +44,15 @@ static int ff_resample_common_##TYPE##_neon(ResampleContext *c, void *dest, cons
int dst_index; \ int dst_index; \
int index= c->index; \ int index= c->index; \
int frac= c->frac; \ int frac= c->frac; \
int sample_index = index >> c->phase_shift; \ int sample_index = 0; \
int x4_aligned_filter_length = c->filter_length & ~3; \ int x4_aligned_filter_length = c->filter_length & ~3; \
int x8_aligned_filter_length = c->filter_length & ~7; \ int x8_aligned_filter_length = c->filter_length & ~7; \
\ \
index &= c->phase_mask; \ while (index >= c->phase_count) { \
sample_index++; \
index -= c->phase_count; \
} \
\
for (dst_index = 0; dst_index < n; dst_index++) { \ for (dst_index = 0; dst_index < n; dst_index++) { \
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index; \ FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index; \
\ \
@@ -75,8 +79,11 @@ static int ff_resample_common_##TYPE##_neon(ResampleContext *c, void *dest, cons
frac -= c->src_incr; \ frac -= c->src_incr; \
index++; \ index++; \
} \ } \
sample_index += index >> c->phase_shift; \ \
index &= c->phase_mask; \ while (index >= c->phase_count) { \
sample_index++; \
index -= c->phase_count; \
} \
} \ } \
\ \
if(update_ctx){ \ if(update_ctx){ \

1
libswresample/options.c
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@@ -85,6 +85,7 @@ static const AVOption options[]={
{"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM }, {"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM },
{"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 24 , PARAM }, {"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 24 , PARAM },
{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_BOOL , {.i64=0 }, 0 , 1 , PARAM }, {"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_BOOL , {.i64=0 }, 0 , 1 , PARAM },
{"exact_rational" , "enable exact rational" , OFFSET(exact_rational) , AV_OPT_TYPE_BOOL , {.i64=0 }, 0 , 1 , PARAM },
{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM }, {"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
/* duplicate option in order to work with avconv */ /* duplicate option in order to work with avconv */

41
libswresample/resample.c
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@@ -297,13 +297,28 @@ fail:
static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta,
double precision, int cheby) double precision, int cheby, int exact_rational)
{ {
double cutoff = cutoff0? cutoff0 : 0.97; double cutoff = cutoff0? cutoff0 : 0.97;
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift; int phase_count= 1<<phase_shift;
if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor if (exact_rational) {
int phase_count_exact, phase_count_exact_den;
av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX);
/* FIXME this is not required, but build_filter needs even phase_count */
if (phase_count_exact & 1 && phase_count_exact > 1 && phase_count_exact < INT_MAX/2)
phase_count_exact *= 2;
if (phase_count_exact <= phase_count) {
/* FIXME this is not required when soft compensation is disabled */
phase_count_exact *= phase_count / phase_count_exact;
phase_count = phase_count_exact;
}
}
if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor
|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format
|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
c = av_mallocz(sizeof(*c)); c = av_mallocz(sizeof(*c));
@@ -337,6 +352,7 @@ static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_r
c->phase_shift = phase_shift; c->phase_shift = phase_shift;
c->phase_mask = phase_count - 1; c->phase_mask = phase_count - 1;
c->phase_count = phase_count;
c->linear = linear; c->linear = linear;
c->factor = factor; c->factor = factor;
c->filter_length = FFMAX((int)ceil(filter_size/factor), 1); c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
@@ -399,7 +415,7 @@ static int swri_resample(ResampleContext *c,
uint8_t *dst, const uint8_t *src, int *consumed, uint8_t *dst, const uint8_t *src, int *consumed,
int src_size, int dst_size, int update_ctx) int src_size, int dst_size, int update_ctx)
{ {
if (c->filter_length == 1 && c->phase_shift == 0) { if (c->filter_length == 1 && c->phase_count == 1) {
int index= c->index; int index= c->index;
int frac= c->frac; int frac= c->frac;
int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index; int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index;
@@ -418,7 +434,7 @@ static int swri_resample(ResampleContext *c,
c->index = 0; c->index = 0;
} }
} else { } else {
int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift; int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;
int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac; int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr; int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
@@ -438,7 +454,7 @@ static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, A
int av_unused mm_flags = av_get_cpu_flags(); int av_unused mm_flags = av_get_cpu_flags();
int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 && int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 &&
(mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2; (mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2;
int64_t max_src_size = (INT64_MAX >> (c->phase_shift+1)) / c->src_incr; int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;
if (c->compensation_distance) if (c->compensation_distance)
dst_size = FFMIN(dst_size, c->compensation_distance); dst_size = FFMIN(dst_size, c->compensation_distance);
@@ -466,11 +482,11 @@ static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, A
static int64_t get_delay(struct SwrContext *s, int64_t base){ static int64_t get_delay(struct SwrContext *s, int64_t base){
ResampleContext *c = s->resample; ResampleContext *c = s->resample;
int64_t num = s->in_buffer_count - (c->filter_length-1)/2; int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
num *= 1 << c->phase_shift; num *= c->phase_count;
num -= c->index; num -= c->index;
num *= c->src_incr; num *= c->src_incr;
num -= c->frac; num -= c->frac;
return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr * c->phase_count);
} }
static int64_t get_out_samples(struct SwrContext *s, int in_samples) { static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
@@ -479,9 +495,9 @@ static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
// They also make it easier to proof that changes and optimizations do not // They also make it easier to proof that changes and optimizations do not
// break the upper bound. // break the upper bound.
int64_t num = s->in_buffer_count + 2LL + in_samples; int64_t num = s->in_buffer_count + 2LL + in_samples;
num *= 1 << c->phase_shift; num *= c->phase_count;
num -= c->index; num -= c->index;
num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) << c->phase_shift, AV_ROUND_UP) + 2; num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) * c->phase_count, AV_ROUND_UP) + 2;
if (c->compensation_distance) { if (c->compensation_distance) {
if (num > INT_MAX) if (num > INT_MAX)
@@ -545,10 +561,13 @@ static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const Audio
} }
res = num - *out_sz; res = num - *out_sz;
*out_idx = c->filter_length + (c->index >> c->phase_shift); *out_idx = c->filter_length;
while (c->index < 0) {
--*out_idx;
c->index += c->phase_count;
}
*out_sz = FFMAX(*out_sz + c->filter_length, *out_sz = FFMAX(*out_sz + c->filter_length,
1 + c->filter_length * 2) - *out_idx; 1 + c->filter_length * 2) - *out_idx;
c->index &= c->phase_mask;
return FFMAX(res, 0); return FFMAX(res, 0);
} }

6
libswresample/resample.h
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@@ -40,8 +40,10 @@ typedef struct ResampleContext {
int frac; int frac;
int src_incr; int src_incr;
int compensation_distance; int compensation_distance;
int phase_shift; /* TODO remove phase_shift and phase_mask */
int phase_mask; attribute_deprecated int phase_shift;
attribute_deprecated int phase_mask;
int phase_count;
int linear; int linear;
enum SwrFilterType filter_type; enum SwrFilterType filter_type;
double kaiser_beta; double kaiser_beta;

30
libswresample/resample_template.c
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@@ -92,9 +92,13 @@ static int RENAME(resample_common)(ResampleContext *c,
int dst_index; int dst_index;
int index= c->index; int index= c->index;
int frac= c->frac; int frac= c->frac;
int sample_index = index >> c->phase_shift; int sample_index = 0;
while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
index &= c->phase_mask;
for (dst_index = 0; dst_index < n; dst_index++) { for (dst_index = 0; dst_index < n; dst_index++) {
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index; FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index;
@@ -111,8 +115,11 @@ static int RENAME(resample_common)(ResampleContext *c,
frac -= c->src_incr; frac -= c->src_incr;
index++; index++;
} }
sample_index += index >> c->phase_shift;
index &= c->phase_mask; while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
} }
if(update_ctx){ if(update_ctx){
@@ -132,12 +139,16 @@ static int RENAME(resample_linear)(ResampleContext *c,
int dst_index; int dst_index;
int index= c->index; int index= c->index;
int frac= c->frac; int frac= c->frac;
int sample_index = index >> c->phase_shift; int sample_index = 0;
#if FILTER_SHIFT == 0 #if FILTER_SHIFT == 0
double inv_src_incr = 1.0 / c->src_incr; double inv_src_incr = 1.0 / c->src_incr;
#endif #endif
index &= c->phase_mask; while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
for (dst_index = 0; dst_index < n; dst_index++) { for (dst_index = 0; dst_index < n; dst_index++) {
FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index; FELEM *filter = ((FELEM *) c->filter_bank) + c->filter_alloc * index;
FELEM2 val=0, v2 = 0; FELEM2 val=0, v2 = 0;
@@ -164,8 +175,11 @@ static int RENAME(resample_linear)(ResampleContext *c,
frac -= c->src_incr; frac -= c->src_incr;
index++; index++;
} }
sample_index += index >> c->phase_shift;
index &= c->phase_mask; while (index >= c->phase_count) {
sample_index++;
index -= c->phase_count;
}
} }
if(update_ctx){ if(update_ctx){

2
libswresample/soxr_resample.c
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@@ -30,7 +30,7 @@
#include <soxr.h> #include <soxr.h>
static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby){ double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational){
soxr_error_t error; soxr_error_t error;
soxr_datatype_t type = soxr_datatype_t type =

2
libswresample/swresample.c
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@@ -262,7 +262,7 @@ av_cold int swr_init(struct SwrContext *s){
} }
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby); s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby, s->exact_rational);
if (!s->resample) { if (!s->resample) {
av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n"); av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
return AVERROR(ENOMEM); return AVERROR(ENOMEM);

3
libswresample/swresample_internal.h
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@@ -69,7 +69,7 @@ struct DitherContext {
}; };
typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby); double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational);
typedef void (* resample_free_func)(struct ResampleContext **c); typedef void (* resample_free_func)(struct ResampleContext **c);
typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
typedef int (* resample_flush_func)(struct SwrContext *c); typedef int (* resample_flush_func)(struct SwrContext *c);
@@ -126,6 +126,7 @@ struct SwrContext {
int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */ int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */ int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */ int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
int exact_rational; /**< if 1 then enable non power of 2 phase_count */
double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */ double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
int filter_type; /**< swr resampling filter type */ int filter_type; /**< swr resampling filter type */
double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */

4
libswresample/version.h
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@@ -29,8 +29,8 @@
#include "libavutil/avutil.h" #include "libavutil/avutil.h"
#define LIBSWRESAMPLE_VERSION_MAJOR 2 #define LIBSWRESAMPLE_VERSION_MAJOR 2
#define LIBSWRESAMPLE_VERSION_MINOR 0 #define LIBSWRESAMPLE_VERSION_MINOR 1
#define LIBSWRESAMPLE_VERSION_MICRO 101 #define LIBSWRESAMPLE_VERSION_MICRO 100
#define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \ #define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \ LIBSWRESAMPLE_VERSION_MINOR, \

4
libswresample/x86/resample_init.c
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@@ -47,6 +47,10 @@ av_cold void swri_resample_dsp_x86_init(ResampleContext *c)
{ {
int av_unused mm_flags = av_get_cpu_flags(); int av_unused mm_flags = av_get_cpu_flags();
/* FIXME use phase_count on asm */
if (c->phase_count != 1 << c->phase_shift)
return;
switch(c->format){ switch(c->format){
case AV_SAMPLE_FMT_S16P: case AV_SAMPLE_FMT_S16P:
if (ARCH_X86_32 && EXTERNAL_MMXEXT(mm_flags)) { if (ARCH_X86_32 && EXTERNAL_MMXEXT(mm_flags)) {