#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "internal.h"
+#include "resample.h"
#include "audio_data.h"
-struct ResampleContext {
- AVAudioResampleContext *avr;
- AudioData *buffer;
- uint8_t *filter_bank;
- int filter_length;
- int ideal_dst_incr;
- int dst_incr;
- int index;
- int frac;
- int src_incr;
- int compensation_distance;
- int phase_shift;
- int phase_mask;
- int linear;
- 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"
-/* 0th order modified bessel function of the first kind. */
+/* 0th order modified Bessel function of the first kind. */
static double bessel(double x)
{
double v = 1;
}
/* Build a polyphase filterbank. */
-static int build_filter(ResampleContext *c)
+static int build_filter(ResampleContext *c, double factor)
{
int ph, i;
- double x, y, w, factor;
+ double x, y, w;
double *tab;
int tap_count = c->filter_length;
int phase_count = 1 << c->phase_shift;
if (!tab)
return AVERROR(ENOMEM);
- /* if upsampling, only need to interpolate, no filter */
- factor = FFMIN(c->factor, 1.0);
-
for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
for (i = 0; i < tap_count; i++) {
c->phase_shift = avr->phase_shift;
c->phase_mask = phase_count - 1;
c->linear = avr->linear_interp;
- c->factor = factor;
c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
c->filter_type = avr->filter_type;
c->kaiser_beta = avr->kaiser_beta;
switch (avr->internal_sample_fmt) {
case AV_SAMPLE_FMT_DBLP:
- c->resample_one = resample_one_dbl;
+ c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl;
+ c->resample_nearest = resample_nearest_dbl;
c->set_filter = set_filter_dbl;
break;
case AV_SAMPLE_FMT_FLTP:
- c->resample_one = resample_one_flt;
+ c->resample_one = c->linear ? resample_linear_flt : resample_one_flt;
+ c->resample_nearest = resample_nearest_flt;
c->set_filter = set_filter_flt;
break;
case AV_SAMPLE_FMT_S32P:
- c->resample_one = resample_one_s32;
+ c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32;
+ c->resample_nearest = resample_nearest_s32;
c->set_filter = set_filter_s32;
break;
case AV_SAMPLE_FMT_S16P:
- c->resample_one = resample_one_s16;
+ c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16;
+ c->resample_nearest = resample_nearest_s16;
c->set_filter = set_filter_s16;
break;
}
+ if (ARCH_AARCH64)
+ ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
+ if (ARCH_ARM)
+ ff_audio_resample_init_arm(c, avr->internal_sample_fmt);
+
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) < 0)
+ if (build_filter(c, factor) < 0)
goto error;
memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
goto error;
c->ideal_dst_incr = c->dst_incr;
- c->index = -phase_count * ((c->filter_length - 1) / 2);
+ c->padding_size = (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),
int compensation_distance)
{
ResampleContext *c;
- AudioData *fifo_buf = NULL;
- int ret = 0;
if (compensation_distance < 0)
return AVERROR(EINVAL);
if (!compensation_distance && sample_delta)
return AVERROR(EINVAL);
- /* if resampling was not enabled previously, re-initialize the
- AVAudioResampleContext and force resampling */
if (!avr->resample_needed) {
- int fifo_samples;
- double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
-
- /* buffer any remaining samples in the output FIFO before closing */
- fifo_samples = av_audio_fifo_size(avr->out_fifo);
- if (fifo_samples > 0) {
- fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
- avr->out_sample_fmt, NULL);
- if (!fifo_buf)
- return AVERROR(EINVAL);
- ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
- fifo_samples);
- if (ret < 0)
- goto reinit_fail;
- }
- /* save the channel mixing matrix */
- ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
- if (ret < 0)
- goto reinit_fail;
-
- /* close the AVAudioResampleContext */
- avresample_close(avr);
-
- avr->force_resampling = 1;
-
- /* restore the channel mixing matrix */
- ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
- if (ret < 0)
- goto reinit_fail;
-
- /* re-open the AVAudioResampleContext */
- ret = avresample_open(avr);
- if (ret < 0)
- goto reinit_fail;
-
- /* restore buffered samples to the output FIFO */
- if (fifo_samples > 0) {
- ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
- fifo_samples);
- if (ret < 0)
- goto reinit_fail;
- ff_audio_data_free(&fifo_buf);
- }
+ av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
+ return AVERROR(EINVAL);
}
c = avr->resample;
c->compensation_distance = compensation_distance;
} else {
c->dst_incr = c->ideal_dst_incr;
}
- return 0;
-reinit_fail:
- ff_audio_data_free(&fifo_buf);
- return ret;
+ return 0;
}
static int resample(ResampleContext *c, void *dst, const void *src,
- int *consumed, int src_size, int dst_size, int update_ctx)
+ int *consumed, int src_size, int dst_size, int update_ctx,
+ 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;
if (!dst != !src)
return AVERROR(EINVAL);
- if (compensation_distance == 0 && c->filter_length == 1 &&
- c->phase_shift == 0) {
- int64_t index2 = ((int64_t)index) << 32;
+ if (nearest_neighbour) {
+ 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 /
if (dst) {
for(dst_index = 0; dst_index < dst_size; dst_index++) {
- c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
+ c->resample_nearest(dst, dst_index, src, index2 >> 32);
index2 += incr;
}
} else {
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, 0, 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;
}
}
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;
return dst_index;
}
-int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
- int *consumed)
+int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
{
- int ch, in_samples, in_leftover, out_samples = 0;
+ int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
int ret = AVERROR(EINVAL);
+ int nearest_neighbour = (c->compensation_distance == 0 &&
+ c->filter_length == 1 &&
+ c->phase_shift == 0);
in_samples = src ? src->nb_samples : 0;
in_leftover = c->buffer->nb_samples;
ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
if (ret < 0)
return ret;
- } else if (!in_leftover) {
+ } else if (in_leftover <= c->final_padding_samples) {
/* no remaining samples to flush */
return 0;
- } else {
- /* 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 (src && 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++) {
+ if (c->buffer->nb_samples > 2 * c->padding_size - i) {
+ memcpy(c->buffer->data[ch] + bps * i,
+ c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
+ } else {
+ memset(c->buffer->data[ch] + bps * i, 0, bps);
+ }
+ }
+ c->initial_padding_filled = 1;
+ }
+
+ if (!src && !c->final_padding_filled) {
+ int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
+ int i;
+
+ ret = ff_audio_data_realloc(c->buffer,
+ FFMAX(in_samples, in_leftover) +
+ c->padding_size);
+ if (ret < 0) {
+ av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
+ return AVERROR(ENOMEM);
+ }
+
+ for (i = 0; i < c->padding_size; i++)
+ for (ch = 0; ch < c->buffer->channels; ch++) {
+ if (in_leftover > i) {
+ memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
+ c->buffer->data[ch] + bps * (in_leftover - i - 1),
+ bps);
+ } else {
+ memset(c->buffer->data[ch] + bps * (in_leftover + i),
+ 0, bps);
+ }
+ }
+ c->buffer->nb_samples += c->padding_size;
+ c->final_padding_samples = c->padding_size;
+ c->final_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) {
out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
- INT_MAX, 0);
+ INT_MAX, 0, nearest_neighbour);
ret = ff_audio_data_realloc(dst, out_samples);
if (ret < 0) {
av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
/* resample each channel plane */
for (ch = 0; ch < c->buffer->channels; ch++) {
out_samples = resample(c, (void *)dst->data[ch],
- (const void *)c->buffer->data[ch], consumed,
+ (const void *)c->buffer->data[ch], &consumed,
c->buffer->nb_samples, dst->allocated_samples,
- ch + 1 == c->buffer->channels);
+ ch + 1 == c->buffer->channels, nearest_neighbour);
}
if (out_samples < 0) {
av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
}
/* drain consumed samples from the internal buffer */
- ff_audio_data_drain(c->buffer, *consumed);
+ 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",
+ av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
in_samples, in_leftover, out_samples, c->buffer->nb_samples);
dst->nb_samples = out_samples;
int avresample_get_delay(AVAudioResampleContext *avr)
{
+ ResampleContext *c = avr->resample;
+
if (!avr->resample_needed || !avr->resample)
return 0;
- return avr->resample->buffer->nb_samples;
+ return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
}
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