[FFmpeg-cvslog] swr: support float & int32 in the resampler
Michael Niedermayer
git at videolan.org
Tue Apr 10 13:33:22 CEST 2012
ffmpeg | branch: master | Michael Niedermayer <michaelni at gmx.at> | Tue Apr 10 13:18:49 2012 +0200| [7f1ae79d38c4edba9dbd31d7bf797e525298ac55] | committer: Michael Niedermayer
swr: support float & int32 in the resampler
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7f1ae79d38c4edba9dbd31d7bf797e525298ac55
---
libswresample/resample.c | 228 ++++++++++++++++-------------------
libswresample/resample_template.c | 113 +++++++++++++++++
libswresample/swresample.c | 2 +-
libswresample/swresample_internal.h | 6 +-
4 files changed, 220 insertions(+), 129 deletions(-)
diff --git a/libswresample/resample.c b/libswresample/resample.c
index d144239..d158e76 100644
--- a/libswresample/resample.c
+++ b/libswresample/resample.c
@@ -26,39 +26,16 @@
*/
#include "libavutil/log.h"
+#include "libavutil/avassert.h"
#include "swresample_internal.h"
-#ifndef CONFIG_RESAMPLE_HP
-#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
#define WINDOW_TYPE 9
-#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
-#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
-#define WINDOW_TYPE 12
-#else
-#define FILTER_SHIFT 0
-#define FELEM double
-#define FELEM2 double
-#define FELEML double
-#define WINDOW_TYPE 24
-#endif
typedef struct ResampleContext {
const AVClass *av_class;
- FELEM *filter_bank;
+ uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
@@ -70,6 +47,9 @@ typedef struct ResampleContext {
int phase_mask;
int linear;
double factor;
+ enum AVSampleFormat format;
+ int felem_size;
+ int filter_shift;
} ResampleContext;
/**
@@ -109,7 +89,7 @@ static double bessel(double x){
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
* @return 0 on success, negative on error
*/
-static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
+static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i;
double x, y, w;
double *tab = av_malloc(tap_count * sizeof(*tab));
@@ -150,12 +130,19 @@ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_c
}
/* normalize so that an uniform color remains the same */
- for(i=0;i<tap_count;i++) {
-#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
- 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
+ switch(c->format){
+ case AV_SAMPLE_FMT_S16:
+ for(i=0;i<tap_count;i++)
+ ((int16_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
+ break;
+ case AV_SAMPLE_FMT_S32:
+ for(i=0;i<tap_count;i++)
+ ((int32_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
+ break;
+ case AV_SAMPLE_FMT_FLT:
+ for(i=0;i<tap_count;i++)
+ ((float*)filter)[ph * tap_count + i] = tab[i] * scale / norm;
+ break;
}
}
#if 0
@@ -199,28 +186,48 @@ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_c
return 0;
}
-ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
+ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat format){
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
- || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
+ || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format) {
c = av_mallocz(sizeof(*c));
if (!c)
return NULL;
+ c->format= format;
+
+ switch(c->format){
+ case AV_SAMPLE_FMT_S16:
+ c->felem_size = 2;
+ c->filter_shift = 15;
+ break;
+ case AV_SAMPLE_FMT_S32:
+ c->felem_size = 4;
+ c->filter_shift = 30;
+ break;
+ case AV_SAMPLE_FMT_FLT:
+ c->felem_size = 4;
+ c->filter_shift = 0;
+ break;
+ default:
+ av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
+ return NULL;
+ }
+
c->phase_shift = phase_shift;
c->phase_mask = phase_count - 1;
c->linear = linear;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
- c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
+ c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*c->felem_size);
if (!c->filter_bank)
goto error;
- if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
+ if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, phase_count, 1<<c->filter_shift, WINDOW_TYPE))
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)*c->felem_size, c->filter_bank, (c->filter_length-1)*c->felem_size);
+ memcpy(c->filter_bank + (c->filter_length*phase_count )*c->felem_size, c->filter_bank + (c->filter_length - 1)*c->felem_size, c->felem_size);
}
c->compensation_distance= 0;
@@ -268,100 +275,69 @@ int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensatio
return 0;
}
-int swri_resample(ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx){
- int dst_index, i;
- int index= c->index;
- int frac= c->frac;
- int dst_incr_frac= c->dst_incr % c->src_incr;
- int dst_incr= c->dst_incr / c->src_incr;
- int compensation_distance= c->compensation_distance;
-
- if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
- int64_t index2= ((int64_t)index)<<32;
- int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
- dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
-
- for(dst_index=0; dst_index < dst_size; dst_index++){
- dst[dst_index] = src[index2>>32];
- index2 += incr;
- }
- index += dst_index * dst_incr;
- index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
- 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;
- FELEM2 val=0;
-
- if(sample_index + c->filter_length > src_size || -sample_index >= src_size){
- break;
- }else if(sample_index < 0){
- for(i=0; i<c->filter_length; i++)
- val += src[FFABS(sample_index + i)] * filter[i];
- }else if(c->linear){
- FELEM2 v2=0;
- for(i=0; i<c->filter_length; i++){
- val += src[sample_index + i] * (FELEM2)filter[i];
- v2 += src[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_AUDIOPHILE_KIDDY_MODE
- dst[dst_index] = av_clip_int16(lrintf(val));
-#else
- val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
- dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
-#endif
-
- frac += dst_incr_frac;
- index += dst_incr;
- if(frac >= c->src_incr){
- frac -= c->src_incr;
- index++;
- }
-
- if(dst_index + 1 == compensation_distance){
- compensation_distance= 0;
- dst_incr_frac= c->ideal_dst_incr % c->src_incr;
- dst_incr= c->ideal_dst_incr / c->src_incr;
- }
- }
- }
- *consumed= FFMAX(index, 0) >> c->phase_shift;
- if(index>=0) index &= c->phase_mask;
-
- if(compensation_distance){
- compensation_distance -= dst_index;
- assert(compensation_distance > 0);
- }
- if(update_ctx){
- c->frac= frac;
- c->index= index;
- c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
- c->compensation_distance= compensation_distance;
- }
-#if 0
- if(update_ctx && !c->compensation_distance){
-#undef rand
- av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
-av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
- }
-#endif
+#define RENAME(N) N ## _int16
+#define FILTER_SHIFT 15
+#define DELEM int16_t
+#define FELEM int16_t
+#define FELEM2 int32_t
+#define FELEML int64_t
+#define FELEM_MAX INT16_MAX
+#define FELEM_MIN INT16_MIN
+#define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\
+ d = (unsigned)(v + 32768) > 65535 ? (v>>31) ^ 32767 : v
+#include "resample_template.c"
+
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+
+#define RENAME(N) N ## _int32
+#define FILTER_SHIFT 30
+#define DELEM int32_t
+#define FELEM int32_t
+#define FELEM2 int64_t
+#define FELEML int64_t
+#define FELEM_MAX INT32_MAX
+#define FELEM_MIN INT32_MIN
+#define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\
+ d = (uint64_t)(v + 0x80000000) > 0xFFFFFFFF ? (v>>63) ^ 0x7FFFFFFF : v
+#include "resample_template.c"
+
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+
+#define RENAME(N) N ## _float
+#define FILTER_SHIFT 0
+#define DELEM float
+#define FELEM float
+#define FELEM2 float
+#define FELEML float
+#define OUT(d, v) d = v
+#include "resample_template.c"
- return dst_index;
-}
int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i, ret= -1;
for(i=0; i<dst->ch_count; i++){
- ret= swri_resample(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
+ if(c->format == AV_SAMPLE_FMT_S16) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
+ if(c->format == AV_SAMPLE_FMT_S32) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
+ if(c->format == AV_SAMPLE_FMT_FLT) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
}
return ret;
diff --git a/libswresample/resample_template.c b/libswresample/resample_template.c
new file mode 100644
index 0000000..5d49374
--- /dev/null
+++ b/libswresample/resample_template.c
@@ -0,0 +1,113 @@
+/*
+ * audio resampling
+ * Copyright (c) 2004-2012 Michael Niedermayer <michaelni at gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio resampling
+ * @author Michael Niedermayer <michaelni at gmx.at>
+ */
+
+int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int *consumed, int src_size, int dst_size, int update_ctx){
+ int dst_index, i;
+ int index= c->index;
+ int frac= c->frac;
+ int dst_incr_frac= c->dst_incr % c->src_incr;
+ int dst_incr= c->dst_incr / c->src_incr;
+ int compensation_distance= c->compensation_distance;
+
+ av_assert1(c->filter_shift == FILTER_SHIFT);
+ av_assert1(c->felem_size == sizeof(FELEM));
+
+ if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
+ int64_t index2= ((int64_t)index)<<32;
+ int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
+ dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
+
+ for(dst_index=0; dst_index < dst_size; dst_index++){
+ dst[dst_index] = src[index2>>32];
+ index2 += incr;
+ }
+ index += dst_index * dst_incr;
+ index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
+ 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= ((FELEM*)c->filter_bank) + c->filter_length*(index & c->phase_mask);
+ int sample_index= index >> c->phase_shift;
+ FELEM2 val=0;
+
+ if(sample_index + c->filter_length > src_size || -sample_index >= src_size){
+ break;
+ }else if(sample_index < 0){
+ for(i=0; i<c->filter_length; i++)
+ val += src[FFABS(sample_index + i)] * filter[i];
+ }else if(c->linear){
+ FELEM2 v2=0;
+ for(i=0; i<c->filter_length; i++){
+ val += src[sample_index + i] * (FELEM2)filter[i];
+ v2 += src[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);
+
+ frac += dst_incr_frac;
+ index += dst_incr;
+ if(frac >= c->src_incr){
+ frac -= c->src_incr;
+ index++;
+ }
+
+ if(dst_index + 1 == compensation_distance){
+ compensation_distance= 0;
+ dst_incr_frac= c->ideal_dst_incr % c->src_incr;
+ dst_incr= c->ideal_dst_incr / c->src_incr;
+ }
+ }
+ }
+ *consumed= FFMAX(index, 0) >> c->phase_shift;
+ if(index>=0) index &= c->phase_mask;
+
+ if(compensation_distance){
+ compensation_distance -= dst_index;
+ assert(compensation_distance > 0);
+ }
+ if(update_ctx){
+ c->frac= frac;
+ c->index= index;
+ c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
+ c->compensation_distance= compensation_distance;
+ }
+#if 0
+ if(update_ctx && !c->compensation_distance){
+#undef rand
+ av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
+av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
+ }
+#endif
+
+ return dst_index;
+}
diff --git a/libswresample/swresample.c b/libswresample/swresample.c
index 205e436..7fd747b 100644
--- a/libswresample/swresample.c
+++ b/libswresample/swresample.c
@@ -190,7 +190,7 @@ int swr_init(struct SwrContext *s){
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
- s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
+ s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8, s->int_sample_fmt);
}else
swri_resample_free(&s->resample);
if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h
index dc7304c..f9f490a 100644
--- a/libswresample/swresample_internal.h
+++ b/libswresample/swresample_internal.h
@@ -78,11 +78,13 @@ struct SwrContext {
/* TODO: callbacks for ASM optimizations */
};
-struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff);
+struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat);
void swri_resample_free(struct ResampleContext **c);
int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
-int swri_resample(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
+int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
+int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
+int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx);
int swri_rematrix_init(SwrContext *s);
int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
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