[FFmpeg-cvslog] resample: add initial padding explicitly
Anton Khirnov
git at videolan.org
Fri Apr 11 20:39:40 CEST 2014
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Tue Mar 4 16:56:01 2014 +0100| [be394968c81019887ef996a78a526bdd85d1e216] | committer: Anton Khirnov
resample: add initial padding explicitly
This simplifies the code, since we do not have to deal with a possibly
negative source index anymore.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=be394968c81019887ef996a78a526bdd85d1e216
---
libavresample/resample.c | 46 ++++++++++++++++++++++++++-----------
libavresample/resample_template.c | 12 ++++------
2 files changed, 36 insertions(+), 22 deletions(-)
diff --git a/libavresample/resample.c b/libavresample/resample.c
index 9048912..c02bba4 100644
--- a/libavresample/resample.c
+++ b/libavresample/resample.c
@@ -33,7 +33,7 @@ struct ResampleContext {
int filter_length;
int ideal_dst_incr;
int dst_incr;
- int index;
+ unsigned int index;
int frac;
int src_incr;
int compensation_distance;
@@ -45,11 +45,13 @@ struct ResampleContext {
double factor;
void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
void (*resample_one)(struct ResampleContext *c, void *dst0,
- int dst_index, const void *src0, int src_size,
- int index, int frac);
+ int dst_index, const void *src0,
+ unsigned int index, int frac);
void (*resample_nearest)(void *dst0, int dst_index,
- const void *src0, int index);
+ const void *src0, unsigned int index);
int padding_size;
+ int initial_padding_filled;
+ int initial_padding_samples;
};
@@ -220,15 +222,18 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
c->ideal_dst_incr = c->dst_incr;
c->padding_size = (c->filter_length - 1) / 2;
- c->index = -phase_count * ((c->filter_length - 1) / 2);
+ c->initial_padding_filled = 0;
+ c->index = 0;
c->frac = 0;
/* allocate internal buffer */
- c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
+ c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
avr->internal_sample_fmt,
"resample buffer");
if (!c->buffer)
goto error;
+ c->buffer->nb_samples = c->padding_size;
+ c->initial_padding_samples = c->padding_size;
av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
av_get_sample_fmt_name(avr->internal_sample_fmt),
@@ -342,7 +347,7 @@ static int resample(ResampleContext *c, void *dst, const void *src,
int nearest_neighbour)
{
int dst_index;
- int index = c->index;
+ unsigned 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;
@@ -352,7 +357,7 @@ static int resample(ResampleContext *c, void *dst, const void *src,
return AVERROR(EINVAL);
if (nearest_neighbour) {
- int64_t index2 = ((int64_t)index) << 32;
+ uint64_t index2 = ((uint64_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 /
@@ -373,12 +378,11 @@ static int resample(ResampleContext *c, void *dst, const void *src,
for (dst_index = 0; dst_index < dst_size; dst_index++) {
int sample_index = index >> c->phase_shift;
- if (sample_index + c->filter_length > src_size ||
- -sample_index >= src_size)
+ if (sample_index + c->filter_length > src_size)
break;
if (dst)
- c->resample_one(c, dst, dst_index, src, src_size, index, frac);
+ c->resample_one(c, dst, dst_index, src, index, frac);
frac += dst_incr_frac;
index += dst_incr;
@@ -394,11 +398,10 @@ static int resample(ResampleContext *c, void *dst, const void *src,
}
}
if (consumed)
- *consumed = FFMAX(index, 0) >> c->phase_shift;
+ *consumed = index >> c->phase_shift;
if (update_ctx) {
- if (index >= 0)
- index &= c->phase_mask;
+ index &= c->phase_mask;
if (compensation_distance) {
compensation_distance -= dst_index;
@@ -437,6 +440,20 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
/* TODO: pad buffer to flush completely */
}
+ if (!c->initial_padding_filled) {
+ int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
+ int i;
+
+ if (c->buffer->nb_samples < 2 * c->padding_size)
+ return 0;
+
+ for (i = 0; i < c->padding_size; i++)
+ for (ch = 0; ch < c->buffer->channels; ch++)
+ memcpy(c->buffer->data[ch] + bps * i,
+ c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
+ c->initial_padding_filled = 1;
+ }
+
/* calculate output size and reallocate output buffer if needed */
/* TODO: try to calculate this without the dummy resample() run */
if (!dst->read_only && dst->allow_realloc) {
@@ -463,6 +480,7 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
/* drain consumed samples from the internal buffer */
ff_audio_data_drain(c->buffer, consumed);
+ c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
in_samples, in_leftover, out_samples, c->buffer->nb_samples);
diff --git a/libavresample/resample_template.c b/libavresample/resample_template.c
index a7cb6ae..661dd0d 100644
--- a/libavresample/resample_template.c
+++ b/libavresample/resample_template.c
@@ -54,7 +54,7 @@
#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
#endif
-static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *src0, int index)
+static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *src0, unsigned int index)
{
FELEM *dst = dst0;
const FELEM *src = src0;
@@ -63,21 +63,17 @@ static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *sr
static void SET_TYPE(resample_one)(ResampleContext *c,
void *dst0, int dst_index, const void *src0,
- int src_size, int index, int frac)
+ unsigned int index, int frac)
{
FELEM *dst = dst0;
const FELEM *src = src0;
int i;
- int sample_index = index >> c->phase_shift;
+ unsigned 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) {
+ if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[sample_index + i] * (FELEM2)filter[i];
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