* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "internal.h"
+#include "resample.h"
#include "audio_data.h"
-#ifdef CONFIG_RESAMPLE_FLT
+
+/* double template */
+#define CONFIG_RESAMPLE_DBL
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_DBL
+
/* float template */
-#define FILTER_SHIFT 0
-#define FELEM float
-#define FELEM2 float
-#define FELEML float
-#define WINDOW_TYPE 24
-#elifdef CONFIG_RESAMPLE_S32
+#define CONFIG_RESAMPLE_FLT
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_FLT
+
/* 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
-#define WINDOW_TYPE 12
-#else
+#define CONFIG_RESAMPLE_S32
+#include "resample_template.c"
+#undef CONFIG_RESAMPLE_S32
+
/* 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
-#define WINDOW_TYPE 9
-#endif
-
-struct ResampleContext {
- AVAudioResampleContext *avr;
- AudioData *buffer;
- FELEM *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;
- double factor;
-};
-
-/**
- * 0th order modified bessel function of the first kind.
- */
+#include "resample_template.c"
+
+
+/* 0th order modified Bessel function of the first kind. */
static double bessel(double x)
{
double v = 1;
return v;
}
-/**
- * Build a polyphase filterbank.
- *
- * @param[out] filter filter coefficients
- * @param factor resampling factor
- * @param tap_count tap count
- * @param phase_count phase count
- * @param scale wanted sum of coefficients for each filter
- * @param type 0->cubic
- * 1->blackman nuttall windowed sinc
- * 2..16->kaiser windowed sinc beta=2..16
- * @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 type)
+/* Build a polyphase filterbank. */
+static int build_filter(ResampleContext *c, double factor)
{
int ph, i;
double x, y, w;
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));
if (!tab)
return AVERROR(ENOMEM);
- /* if upsampling, only need to interpolate, no filter */
- if (factor > 1.0)
- factor = 1.0;
-
for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
for (i = 0; i < tap_count; i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
- switch (type) {
- case 0: {
+ 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);
if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
break;
}
- case 1:
+ case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0 * x / (factor * tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos( w) +
0.1365995 * cos(2 * w) -
0.0106411 * cos(3 * w);
break;
- default:
+ case AV_RESAMPLE_FILTER_TYPE_KAISER:
w = 2.0 * x / (factor * tap_count * M_PI);
- y *= bessel(type * 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);
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));
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 = 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 = 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 = 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 = c->linear ? resample_linear_s16 : resample_one_s16;
+ c->resample_nearest = resample_nearest_s16;
+ c->set_filter = set_filter_s16;
+ break;
+ }
- c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
+ 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->filter_bank, factor, c->filter_length, phase_count,
- 1 << FILTER_SHIFT, WINDOW_TYPE) < 0)
+ if (build_filter(c, factor) < 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,
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, int16_t *dst, const int16_t *src,
- int *consumed, int src_size, int dst_size, int update_ctx)
+static int resample(ResampleContext *c, void *dst, const void *src,
+ int *consumed, int src_size, int dst_size, int update_ctx,
+ int nearest_neighbour)
{
- int dst_index, i;
- int index = c->index;
+ int dst_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++) {
- dst[dst_index] = src[index2 >> 32];
+ c->resample_nearest(dst, dst_index, src, index2 >> 32);
index2 += incr;
}
} else {
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)
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, 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, (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);
+ 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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