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resample: add initial padding explicitly

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This simplifies the code, since we do not have to deal with a possibly
negative source index anymore.
pull/64/head
Anton Khirnov 11 years ago
parent eed752d61d
commit be394968c8
2 changed files with 36 additions and 22 deletions
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  1. 46
      libavresample/resample.c
  2. 12
      libavresample/resample_template.c

46
libavresample/resample.c
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@ -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);

12
libavresample/resample_template.c
Unescape View File

@ -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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