[FFmpeg-cvslog] lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
Justin Ruggles
git at videolan.org
Mon Jul 9 22:43:08 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sun May 27 21:44:55 2012 -0400| [6410397600eae3bd447c0ec2667cc53722ab84ee] | committer: Justin Ruggles
lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
Based partially on implementation by Michael Niedermayer <michaelni at gmx.at> in
libswresample in FFmpeg. See commits:
7f1ae79d38c4edba9dbd31d7bf797e525298ac55
24ab1abfb6d55bf330022df4b10d7aec80b3f116
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=6410397600eae3bd447c0ec2667cc53722ab84ee
---
libavresample/resample.c | 163 +++++++++++++++++--------------------
libavresample/resample_template.c | 102 +++++++++++++++++++++++
libavresample/utils.c | 22 ++++-
3 files changed, 199 insertions(+), 88 deletions(-)
diff --git a/libavresample/resample.c b/libavresample/resample.c
index 7316e4e..1c3d13a 100644
--- a/libavresample/resample.c
+++ b/libavresample/resample.c
@@ -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);
}
diff --git a/libavresample/resample_template.c b/libavresample/resample_template.c
new file mode 100644
index 0000000..5b0fbec
--- /dev/null
+++ b/libavresample/resample_template.c
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni at 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
diff --git a/libavresample/utils.c b/libavresample/utils.c
index ac1c36e..1aca566 100644
--- a/libavresample/utils.c
+++ b/libavresample/utils.c
@@ -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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