[FFmpeg-cvslog] avfilter/af_afir: add support for double sample format
Paul B Mahol
git at videolan.org
Sun May 15 14:33:16 EEST 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat May 14 10:28:49 2022 +0200| [163e737c1793eeea9c2df15298253ffc04906afe] | committer: Paul B Mahol
avfilter/af_afir: add support for double sample format
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=163e737c1793eeea9c2df15298253ffc04906afe
---
doc/filters.texi | 16 ++
libavfilter/af_afir.c | 511 +++++++++-----------------------------------
libavfilter/af_afir.h | 99 +++++++++
libavfilter/af_afirdsp.h | 20 ++
libavfilter/afir_template.c | 392 +++++++++++++++++++++++++++++++++
5 files changed, 623 insertions(+), 415 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index b6cbdc85cc..c243caee20 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1655,6 +1655,22 @@ Allowed range is from @var{1} to @var{32}. Default is @var{1}.
Set IR stream which will be used for convolution, starting from @var{0}, should always be
lower than supplied value by @code{nbirs} option. Default is @var{0}.
This option can be changed at runtime via @ref{commands}.
+
+ at item precision
+Set which precision to use when processing samples.
+
+ at table @option
+ at item auto
+Auto pick internal sample format depending on other filters.
+
+ at item float
+Always use single-floating point precision sample format.
+
+ at item double
+Always use double-floating point precision sample format.
+ at end table
+
+Default value is auto.
@end table
@subsection Examples
diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
index 301553575f..e1fe7d6a64 100644
--- a/libavfilter/af_afir.c
+++ b/libavfilter/af_afir.c
@@ -42,208 +42,78 @@
#include "filters.h"
#include "formats.h"
#include "internal.h"
+#include "af_afir.h"
#include "af_afirdsp.h"
-typedef struct AudioFIRSegment {
- int nb_partitions;
- int part_size;
- int block_size;
- int fft_length;
- int coeff_size;
- int input_size;
- int input_offset;
-
- int *output_offset;
- int *part_index;
-
- AVFrame *sumin;
- AVFrame *sumout;
- AVFrame *blockin;
- AVFrame *blockout;
- AVFrame *buffer;
- AVFrame *coeff;
- AVFrame *input;
- AVFrame *output;
-
- AVTXContext **tx, **itx;
- av_tx_fn tx_fn, itx_fn;
-} AudioFIRSegment;
-
-typedef struct AudioFIRContext {
- const AVClass *class;
-
- float wet_gain;
- float dry_gain;
- float length;
- int gtype;
- float ir_gain;
- int ir_format;
- float max_ir_len;
- int response;
- int w, h;
- AVRational frame_rate;
- int ir_channel;
- int minp;
- int maxp;
- int nb_irs;
- int selir;
-
- float gain;
-
- int eof_coeffs[32];
- int have_coeffs;
- int nb_taps;
- int nb_channels;
- int nb_coef_channels;
- int one2many;
-
- AudioFIRSegment seg[1024];
- int nb_segments;
-
- AVFrame *in;
- AVFrame *ir[32];
- AVFrame *video;
- int min_part_size;
- int64_t pts;
+static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
+{
+ const uint8_t *font;
+ int font_height;
+ int i;
- AudioFIRDSPContext afirdsp;
- AVFloatDSPContext *fdsp;
-} AudioFIRContext;
+ font = avpriv_cga_font, font_height = 8;
-static void direct(const float *in, const AVComplexFloat *ir, int len, float *out)
-{
- for (int n = 0; n < len; n++)
- for (int m = 0; m <= n; m++)
- out[n] += ir[m].re * in[n - m];
-}
+ for (i = 0; txt[i]; i++) {
+ int char_y, mask;
-static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
-{
- if ((nb_samples & 15) == 0 && nb_samples >= 16) {
- s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
- } else {
- for (int n = 0; n < nb_samples; n++)
- dst[n] += src[n];
+ uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
+ for (char_y = 0; char_y < font_height; char_y++) {
+ for (mask = 0x80; mask; mask >>= 1) {
+ if (font[txt[i] * font_height + char_y] & mask)
+ AV_WL32(p, color);
+ p += 4;
+ }
+ p += pic->linesize[0] - 8 * 4;
+ }
}
}
-static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
+static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
{
- AudioFIRContext *s = ctx->priv;
- const float *in = (const float *)s->in->extended_data[ch] + offset;
- float *blockin, *blockout, *buf, *ptr = (float *)out->extended_data[ch] + offset;
- const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
- int n, i, j;
-
- for (int segment = 0; segment < s->nb_segments; segment++) {
- AudioFIRSegment *seg = &s->seg[segment];
- float *src = (float *)seg->input->extended_data[ch];
- float *dst = (float *)seg->output->extended_data[ch];
- float *sumin = (float *)seg->sumin->extended_data[ch];
- float *sumout = (float *)seg->sumout->extended_data[ch];
-
- if (s->min_part_size >= 8) {
- s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
- emms_c();
- } else {
- for (n = 0; n < nb_samples; n++)
- src[seg->input_offset + n] = in[n] * s->dry_gain;
- }
-
- seg->output_offset[ch] += s->min_part_size;
- if (seg->output_offset[ch] == seg->part_size) {
- seg->output_offset[ch] = 0;
- } else {
- memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
-
- dst += seg->output_offset[ch];
- fir_fadd(s, ptr, dst, nb_samples);
- continue;
- }
-
- if (seg->part_size < 8) {
- memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
-
- j = seg->part_index[ch];
-
- for (i = 0; i < seg->nb_partitions; i++) {
- const int coffset = j * seg->coeff_size;
- const AVComplexFloat *coeff = (const AVComplexFloat *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
-
- direct(src, coeff, nb_samples, dst);
+ int dx = FFABS(x1-x0);
+ int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
+ int err = (dx>dy ? dx : -dy) / 2, e2;
- if (j == 0)
- j = seg->nb_partitions;
- j--;
- }
+ for (;;) {
+ AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
- seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+ if (x0 == x1 && y0 == y1)
+ break;
- memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+ e2 = err;
- for (n = 0; n < nb_samples; n++) {
- ptr[n] += dst[n];
- }
- continue;
+ if (e2 >-dx) {
+ err -= dy;
+ x0--;
}
- memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
- blockin = (float *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size;
- blockout = (float *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
- memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size));
-
- memcpy(blockin, src, sizeof(*src) * seg->part_size);
-
- seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(float));
-
- j = seg->part_index[ch];
-
- for (i = 0; i < seg->nb_partitions; i++) {
- const int coffset = j * seg->coeff_size;
- const float *blockout = (const float *)seg->blockout->extended_data[ch] + i * seg->block_size;
- const AVComplexFloat *coeff = (const AVComplexFloat *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
-
- s->afirdsp.fcmul_add(sumin, blockout, (const float *)coeff, seg->part_size);
-
- if (j == 0)
- j = seg->nb_partitions;
- j--;
+ if (e2 < dy) {
+ err += dx;
+ y0 += sy;
}
-
- seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(float));
-
- buf = (float *)seg->buffer->extended_data[ch];
- fir_fadd(s, buf, sumout, seg->part_size);
-
- memcpy(dst, buf, seg->part_size * sizeof(*dst));
-
- buf = (float *)seg->buffer->extended_data[ch];
- memcpy(buf, sumout + seg->part_size, seg->part_size * sizeof(*buf));
-
- seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
-
- memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
-
- fir_fadd(s, ptr, dst, nb_samples);
}
+}
- if (s->min_part_size >= 8) {
- s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
- emms_c();
- } else {
- for (n = 0; n < nb_samples; n++)
- ptr[n] *= s->wet_gain;
- }
+#define DEPTH 32
+#include "afir_template.c"
- return 0;
-}
+#undef DEPTH
+#define DEPTH 64
+#include "afir_template.c"
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
{
AudioFIRContext *s = ctx->priv;
for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
- fir_quantum(ctx, out, ch, offset);
+ switch (s->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ fir_quantum_float(ctx, out, ch, offset);
+ break;
+ case AV_SAMPLE_FMT_DBLP:
+ fir_quantum_double(ctx, out, ch, offset);
+ break;
+ }
}
return 0;
@@ -284,144 +154,6 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
return ff_filter_frame(outlink, out);
}
-static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
-{
- const uint8_t *font;
- int font_height;
- int i;
-
- font = avpriv_cga_font, font_height = 8;
-
- for (i = 0; txt[i]; i++) {
- int char_y, mask;
-
- uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
- for (char_y = 0; char_y < font_height; char_y++) {
- for (mask = 0x80; mask; mask >>= 1) {
- if (font[txt[i] * font_height + char_y] & mask)
- AV_WL32(p, color);
- p += 4;
- }
- p += pic->linesize[0] - 8 * 4;
- }
- }
-}
-
-static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
-{
- int dx = FFABS(x1-x0);
- int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
- int err = (dx>dy ? dx : -dy) / 2, e2;
-
- for (;;) {
- AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
-
- if (x0 == x1 && y0 == y1)
- break;
-
- e2 = err;
-
- if (e2 >-dx) {
- err -= dy;
- x0--;
- }
-
- if (e2 < dy) {
- err += dx;
- y0 += sy;
- }
- }
-}
-
-static void draw_response(AVFilterContext *ctx, AVFrame *out)
-{
- AudioFIRContext *s = ctx->priv;
- float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
- float min_delay = FLT_MAX, max_delay = FLT_MIN;
- int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
- char text[32];
- int channel, i, x;
-
- memset(out->data[0], 0, s->h * out->linesize[0]);
-
- phase = av_malloc_array(s->w, sizeof(*phase));
- mag = av_malloc_array(s->w, sizeof(*mag));
- delay = av_malloc_array(s->w, sizeof(*delay));
- if (!mag || !phase || !delay)
- goto end;
-
- channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1);
- for (i = 0; i < s->w; i++) {
- const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
- double w = i * M_PI / (s->w - 1);
- double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
-
- for (x = 0; x < s->nb_taps; x++) {
- real += cos(-x * w) * src[x];
- imag += sin(-x * w) * src[x];
- real_num += cos(-x * w) * src[x] * x;
- imag_num += sin(-x * w) * src[x] * x;
- }
-
- mag[i] = hypot(real, imag);
- phase[i] = atan2(imag, real);
- div = real * real + imag * imag;
- delay[i] = (real_num * real + imag_num * imag) / div;
- min = fminf(min, mag[i]);
- max = fmaxf(max, mag[i]);
- min_delay = fminf(min_delay, delay[i]);
- max_delay = fmaxf(max_delay, delay[i]);
- }
-
- for (i = 0; i < s->w; i++) {
- int ymag = mag[i] / max * (s->h - 1);
- int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
- int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
-
- ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
- yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
- ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
-
- if (prev_ymag < 0)
- prev_ymag = ymag;
- if (prev_yphase < 0)
- prev_yphase = yphase;
- if (prev_ydelay < 0)
- prev_ydelay = ydelay;
-
- draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
- draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
- draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
-
- prev_ymag = ymag;
- prev_yphase = yphase;
- prev_ydelay = ydelay;
- }
-
- if (s->w > 400 && s->h > 100) {
- drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
- snprintf(text, sizeof(text), "%.2f", max);
- drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
-
- drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
- snprintf(text, sizeof(text), "%.2f", min);
- drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
-
- drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
- snprintf(text, sizeof(text), "%.2f", max_delay);
- drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
-
- drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
- snprintf(text, sizeof(text), "%.2f", min_delay);
- drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
- }
-
-end:
- av_free(delay);
- av_free(phase);
- av_free(mag);
-}
-
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
int offset, int nb_partitions, int part_size)
{
@@ -446,9 +178,20 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
return AVERROR(ENOMEM);
for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 8; ch++) {
- float scale = 1.f, iscale = 1.f / part_size;
- av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_FLOAT_RDFT, 0, 2 * part_size, &scale, 0);
- av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_FLOAT_RDFT, 1, 2 * part_size, &iscale, 0);
+ double dscale = 1.0, idscale = 1.0 / part_size;
+ float fscale = 1.f, ifscale = 1.f / part_size;
+
+ switch (s->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_FLOAT_RDFT, 0, 2 * part_size, &fscale, 0);
+ av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_FLOAT_RDFT, 1, 2 * part_size, &ifscale, 0);
+ break;
+ case AV_SAMPLE_FMT_DBLP:
+ av_tx_init(&seg->tx[ch], &seg->tx_fn, AV_TX_DOUBLE_RDFT, 0, 2 * part_size, &dscale, 0);
+ av_tx_init(&seg->itx[ch], &seg->itx_fn, AV_TX_DOUBLE_RDFT, 1, 2 * part_size, &idscale, 0);
+ break;
+ }
+
if (!seg->tx[ch] || !seg->itx[ch])
return AVERROR(ENOMEM);
}
@@ -502,8 +245,7 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
static int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- int ret, i, ch, n, cur_nb_taps;
- float power = 0;
+ int ret, i, cur_nb_taps;
if (!s->nb_taps) {
int part_size, max_part_size;
@@ -546,109 +288,42 @@ static int convert_coeffs(AVFilterContext *ctx)
return AVERROR_BUG;
}
- if (s->response)
- draw_response(ctx, s->video);
+ if (s->response) {
+ switch (s->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ draw_response_float(ctx, s->video);
+ break;
+ case AV_SAMPLE_FMT_DBLP:
+ draw_response_double(ctx, s->video);
+ break;
+ }
+ }
s->gain = 1;
cur_nb_taps = s->ir[s->selir]->nb_samples;
- switch (s->gtype) {
- case -1:
- /* nothing to do */
+ switch (s->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ ret = get_power_float(ctx, s, cur_nb_taps);
break;
- case 0:
- for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
- float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
-
- for (i = 0; i < cur_nb_taps; i++)
- power += FFABS(time[i]);
- }
- s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
+ case AV_SAMPLE_FMT_DBLP:
+ ret = get_power_double(ctx, s, cur_nb_taps);
break;
- case 1:
- for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
- float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
-
- for (i = 0; i < cur_nb_taps; i++)
- power += time[i];
- }
- s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
- break;
- case 2:
- for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
- float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
-
- for (i = 0; i < cur_nb_taps; i++)
- power += time[i] * time[i];
- }
- s->gain = sqrtf(ch / power);
- break;
- default:
- return AVERROR_BUG;
}
- s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
- av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
- for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
- float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
-
- s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
- }
+ if (ret < 0)
+ return ret;
av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
- for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
- float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- int toffset = 0;
-
- for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
- time[i] = 0;
-
- av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
-
- for (int segment = 0; segment < s->nb_segments; segment++) {
- AudioFIRSegment *seg = &s->seg[segment];
- float *blockin = (float *)seg->blockin->extended_data[ch];
- float *blockout = (float *)seg->blockout->extended_data[ch];
- AVComplexFloat *coeff = (AVComplexFloat *)seg->coeff->extended_data[ch];
-
- av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
-
- for (i = 0; i < seg->nb_partitions; i++) {
- const int coffset = i * seg->coeff_size;
- const int remaining = s->nb_taps - toffset;
- const int size = remaining >= seg->part_size ? seg->part_size : remaining;
-
- if (size < 8) {
- for (n = 0; n < size; n++)
- coeff[coffset + n].re = time[toffset + n];
-
- toffset += size;
- continue;
- }
-
- memset(blockin, 0, sizeof(*blockin) * seg->fft_length);
- memcpy(blockin, time + toffset, size * sizeof(*blockin));
-
- seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(float));
-
- for (n = 0; n < seg->part_size + 1; n++) {
- coeff[coffset + n].re = blockout[2 * n];
- coeff[coffset + n].im = blockout[2 * n + 1];
- }
-
- toffset += size;
- }
-
- av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
- av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
- av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
- av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
- av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
- av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
- av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
- }
+ switch (s->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ convert_channels_float(ctx, s);
+ break;
+ case AV_SAMPLE_FMT_DBLP:
+ convert_channels_double(ctx, s);
+ break;
}
s->have_coeffs = 1;
@@ -762,9 +437,10 @@ static int activate(AVFilterContext *ctx)
static int query_formats(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- static const enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE
+ static const enum AVSampleFormat sample_fmts[3][3] = {
+ { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
+ { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
+ { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
};
static const enum AVPixelFormat pix_fmts[] = {
AV_PIX_FMT_RGB0,
@@ -801,7 +477,7 @@ static int query_formats(AVFilterContext *ctx)
}
}
- if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts)) < 0)
+ if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
@@ -827,6 +503,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
s->nb_channels = outlink->ch_layout.nb_channels;
s->nb_coef_channels = ctx->inputs[1 + s->selir]->ch_layout.nb_channels;
+ s->format = outlink->format;
return 0;
}
@@ -977,6 +654,10 @@ static const AVOption afir_options[] = {
{ "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
{ "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
{ "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
+ { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, "precision" },
+ { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
+ { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
+ { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
{ NULL }
};
diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
new file mode 100644
index 0000000000..e82fc58589
--- /dev/null
+++ b/libavfilter/af_afir.h
@@ -0,0 +1,99 @@
+/*
+ * Copyright (c) 2017 Paul B Mahol
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFILTER_AFIR_H
+#define AVFILTER_AFIR_H
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/frame.h"
+#include "libavutil/rational.h"
+#include "libavutil/tx.h"
+#include "avfilter.h"
+#include "af_afirdsp.h"
+
+typedef struct AudioFIRSegment {
+ int nb_partitions;
+ int part_size;
+ int block_size;
+ int fft_length;
+ int coeff_size;
+ int input_size;
+ int input_offset;
+
+ int *output_offset;
+ int *part_index;
+
+ AVFrame *sumin;
+ AVFrame *sumout;
+ AVFrame *blockin;
+ AVFrame *blockout;
+ AVFrame *buffer;
+ AVFrame *coeff;
+ AVFrame *input;
+ AVFrame *output;
+
+ AVTXContext **tx, **itx;
+ av_tx_fn tx_fn, itx_fn;
+} AudioFIRSegment;
+
+typedef struct AudioFIRContext {
+ const AVClass *class;
+
+ float wet_gain;
+ float dry_gain;
+ float length;
+ int gtype;
+ float ir_gain;
+ int ir_format;
+ float max_ir_len;
+ int response;
+ int w, h;
+ AVRational frame_rate;
+ int ir_channel;
+ int minp;
+ int maxp;
+ int nb_irs;
+ int selir;
+ int precision;
+ int format;
+
+ double gain;
+
+ int eof_coeffs[32];
+ int have_coeffs;
+ int nb_taps;
+ int nb_channels;
+ int nb_coef_channels;
+ int one2many;
+
+ AudioFIRSegment seg[1024];
+ int nb_segments;
+
+ AVFrame *in;
+ AVFrame *ir[32];
+ AVFrame *video;
+ int min_part_size;
+ int64_t pts;
+
+ AudioFIRDSPContext afirdsp;
+ AVFloatDSPContext *fdsp;
+} AudioFIRContext;
+
+#endif /* AVFILTER_AFIR_H */
diff --git a/libavfilter/af_afirdsp.h b/libavfilter/af_afirdsp.h
index 05182bebb4..bf7d1d6f0f 100644
--- a/libavfilter/af_afirdsp.h
+++ b/libavfilter/af_afirdsp.h
@@ -29,6 +29,8 @@
typedef struct AudioFIRDSPContext {
void (*fcmul_add)(float *sum, const float *t, const float *c,
ptrdiff_t len);
+ void (*dcmul_add)(double *sum, const double *t, const double *c,
+ ptrdiff_t len);
} AudioFIRDSPContext;
void ff_afir_init_x86(AudioFIRDSPContext *s);
@@ -50,9 +52,27 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le
sum[2 * n] += t[2 * n] * c[2 * n];
}
+static void dcmul_add_c(double *sum, const double *t, const double *c, ptrdiff_t len)
+{
+ int n;
+
+ for (n = 0; n < len; n++) {
+ const double cre = c[2 * n ];
+ const double cim = c[2 * n + 1];
+ const double tre = t[2 * n ];
+ const double tim = t[2 * n + 1];
+
+ sum[2 * n ] += tre * cre - tim * cim;
+ sum[2 * n + 1] += tre * cim + tim * cre;
+ }
+
+ sum[2 * n] += t[2 * n] * c[2 * n];
+}
+
static av_unused void ff_afir_init(AudioFIRDSPContext *dsp)
{
dsp->fcmul_add = fcmul_add_c;
+ dsp->dcmul_add = dcmul_add_c;
if (ARCH_X86)
ff_afir_init_x86(dsp);
diff --git a/libavfilter/afir_template.c b/libavfilter/afir_template.c
new file mode 100644
index 0000000000..6cb3eb2203
--- /dev/null
+++ b/libavfilter/afir_template.c
@@ -0,0 +1,392 @@
+/*
+ * Copyright (c) 2017 Paul B Mahol
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+#include "audio.h"
+
+#undef ctype
+#undef ftype
+#undef SQRT
+#undef SAMPLE_FORMAT
+#if DEPTH == 32
+#define SAMPLE_FORMAT float
+#define SQRT sqrtf
+#define ctype AVComplexFloat
+#define ftype float
+#else
+#define SAMPLE_FORMAT double
+#define SQRT sqrt
+#define ctype AVComplexDouble
+#define ftype double
+#endif
+
+#define fn3(a,b) a##_##b
+#define fn2(a,b) fn3(a,b)
+#define fn(a) fn2(a, SAMPLE_FORMAT)
+
+static void fn(draw_response)(AVFilterContext *ctx, AVFrame *out)
+{
+ AudioFIRContext *s = ctx->priv;
+ ftype *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
+ ftype min_delay = FLT_MAX, max_delay = FLT_MIN;
+ int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
+ char text[32];
+ int channel, i, x;
+
+ memset(out->data[0], 0, s->h * out->linesize[0]);
+
+ phase = av_malloc_array(s->w, sizeof(*phase));
+ mag = av_malloc_array(s->w, sizeof(*mag));
+ delay = av_malloc_array(s->w, sizeof(*delay));
+ if (!mag || !phase || !delay)
+ goto end;
+
+ channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->ch_layout.nb_channels - 1);
+ for (i = 0; i < s->w; i++) {
+ const ftype *src = (const ftype *)s->ir[s->selir]->extended_data[channel];
+ double w = i * M_PI / (s->w - 1);
+ double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
+
+ for (x = 0; x < s->nb_taps; x++) {
+ real += cos(-x * w) * src[x];
+ imag += sin(-x * w) * src[x];
+ real_num += cos(-x * w) * src[x] * x;
+ imag_num += sin(-x * w) * src[x] * x;
+ }
+
+ mag[i] = hypot(real, imag);
+ phase[i] = atan2(imag, real);
+ div = real * real + imag * imag;
+ delay[i] = (real_num * real + imag_num * imag) / div;
+ min = fminf(min, mag[i]);
+ max = fmaxf(max, mag[i]);
+ min_delay = fminf(min_delay, delay[i]);
+ max_delay = fmaxf(max_delay, delay[i]);
+ }
+
+ for (i = 0; i < s->w; i++) {
+ int ymag = mag[i] / max * (s->h - 1);
+ int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
+ int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
+
+ ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
+ yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
+ ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
+
+ if (prev_ymag < 0)
+ prev_ymag = ymag;
+ if (prev_yphase < 0)
+ prev_yphase = yphase;
+ if (prev_ydelay < 0)
+ prev_ydelay = ydelay;
+
+ draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
+ draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
+ draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
+
+ prev_ymag = ymag;
+ prev_yphase = yphase;
+ prev_ydelay = ydelay;
+ }
+
+ if (s->w > 400 && s->h > 100) {
+ drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
+ snprintf(text, sizeof(text), "%.2f", max);
+ drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
+
+ drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
+ snprintf(text, sizeof(text), "%.2f", min);
+ drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
+
+ drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
+ snprintf(text, sizeof(text), "%.2f", max_delay);
+ drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
+
+ drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
+ snprintf(text, sizeof(text), "%.2f", min_delay);
+ drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
+ }
+
+end:
+ av_free(delay);
+ av_free(phase);
+ av_free(mag);
+}
+
+static void fn(convert_channels)(AVFilterContext *ctx, AudioFIRContext *s)
+{
+ for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
+ ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
+ int toffset = 0;
+
+ for (int i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
+ time[i] = 0;
+
+ av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
+
+ for (int segment = 0; segment < s->nb_segments; segment++) {
+ AudioFIRSegment *seg = &s->seg[segment];
+ ftype *blockin = (ftype *)seg->blockin->extended_data[ch];
+ ftype *blockout = (ftype *)seg->blockout->extended_data[ch];
+ ctype *coeff = (ctype *)seg->coeff->extended_data[ch];
+
+ av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
+
+ for (int i = 0; i < seg->nb_partitions; i++) {
+ const int coffset = i * seg->coeff_size;
+ const int remaining = s->nb_taps - toffset;
+ const int size = remaining >= seg->part_size ? seg->part_size : remaining;
+
+ if (size < 8) {
+ for (int n = 0; n < size; n++)
+ coeff[coffset + n].re = time[toffset + n];
+
+ toffset += size;
+ continue;
+ }
+
+ memset(blockin, 0, sizeof(*blockin) * seg->fft_length);
+ memcpy(blockin, time + toffset, size * sizeof(*blockin));
+
+ seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(ftype));
+
+ for (int n = 0; n < seg->part_size + 1; n++) {
+ coeff[coffset + n].re = blockout[2 * n];
+ coeff[coffset + n].im = blockout[2 * n + 1];
+ }
+
+ toffset += size;
+ }
+
+ av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
+ av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
+ av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
+ av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
+ av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
+ av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
+ av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
+ }
+ }
+}
+
+static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s, int cur_nb_taps)
+{
+ ftype power = 0;
+ int ch;
+
+ switch (s->gtype) {
+ case -1:
+ /* nothing to do */
+ break;
+ case 0:
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
+ ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
+
+ for (int i = 0; i < cur_nb_taps; i++)
+ power += FFABS(time[i]);
+ }
+ s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
+ break;
+ case 1:
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
+ ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
+
+ for (int i = 0; i < cur_nb_taps; i++)
+ power += time[i];
+ }
+ s->gain = ctx->inputs[1 + s->selir]->ch_layout.nb_channels / power;
+ break;
+ case 2:
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
+ ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
+
+ for (int i = 0; i < cur_nb_taps; i++)
+ power += time[i] * time[i];
+ }
+ s->gain = SQRT(ch / power);
+ break;
+ default:
+ return AVERROR_BUG;
+ }
+
+ s->gain = FFMIN(s->gain * s->ir_gain, 1.);
+
+ av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
+
+ for (int ch = 0; ch < ctx->inputs[1 + s->selir]->ch_layout.nb_channels; ch++) {
+ ftype *time = (ftype *)s->ir[s->selir]->extended_data[!s->one2many * ch];
+
+#if DEPTH == 32
+ s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
+#else
+ s->fdsp->vector_dmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 8));
+#endif
+ }
+
+ return 0;
+}
+
+static void fn(direct)(const ftype *in, const ctype *ir, int len, ftype *out)
+{
+ for (int n = 0; n < len; n++)
+ for (int m = 0; m <= n; m++)
+ out[n] += ir[m].re * in[n - m];
+}
+
+static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int nb_samples)
+{
+ if ((nb_samples & 15) == 0 && nb_samples >= 16) {
+#if DEPTH == 32
+ s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
+#else
+ s->fdsp->vector_dmac_scalar(dst, src, 1.0, nb_samples);
+#endif
+ } else {
+ for (int n = 0; n < nb_samples; n++)
+ dst[n] += src[n];
+ }
+}
+
+static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
+{
+ AudioFIRContext *s = ctx->priv;
+ const ftype *in = (const ftype *)s->in->extended_data[ch] + offset;
+ ftype *blockin, *blockout, *buf, *ptr = (ftype *)out->extended_data[ch] + offset;
+ const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
+ int n, i, j;
+
+ for (int segment = 0; segment < s->nb_segments; segment++) {
+ AudioFIRSegment *seg = &s->seg[segment];
+ ftype *src = (ftype *)seg->input->extended_data[ch];
+ ftype *dst = (ftype *)seg->output->extended_data[ch];
+ ftype *sumin = (ftype *)seg->sumin->extended_data[ch];
+ ftype *sumout = (ftype *)seg->sumout->extended_data[ch];
+
+ if (s->min_part_size >= 8) {
+#if DEPTH == 32
+ s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
+#else
+ s->fdsp->vector_dmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 8));
+#endif
+ emms_c();
+ } else {
+ for (n = 0; n < nb_samples; n++)
+ src[seg->input_offset + n] = in[n] * s->dry_gain;
+ }
+
+ seg->output_offset[ch] += s->min_part_size;
+ if (seg->output_offset[ch] == seg->part_size) {
+ seg->output_offset[ch] = 0;
+ } else {
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ dst += seg->output_offset[ch];
+ fn(fir_fadd)(s, ptr, dst, nb_samples);
+ continue;
+ }
+
+ if (seg->part_size < 8) {
+ memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
+
+ j = seg->part_index[ch];
+
+ for (i = 0; i < seg->nb_partitions; i++) {
+ const int coffset = j * seg->coeff_size;
+ const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
+
+ fn(direct)(src, coeff, nb_samples, dst);
+
+ if (j == 0)
+ j = seg->nb_partitions;
+ j--;
+ }
+
+ seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ for (n = 0; n < nb_samples; n++) {
+ ptr[n] += dst[n];
+ }
+ continue;
+ }
+
+ memset(sumin, 0, sizeof(*sumin) * seg->fft_length);
+ blockin = (ftype *)seg->blockin->extended_data[ch] + seg->part_index[ch] * seg->block_size;
+ blockout = (ftype *)seg->blockout->extended_data[ch] + seg->part_index[ch] * seg->block_size;
+ memset(blockin + seg->part_size, 0, sizeof(*blockin) * (seg->fft_length - seg->part_size));
+
+ memcpy(blockin, src, sizeof(*src) * seg->part_size);
+
+ seg->tx_fn(seg->tx[ch], blockout, blockin, sizeof(ftype));
+
+ j = seg->part_index[ch];
+
+ for (i = 0; i < seg->nb_partitions; i++) {
+ const int coffset = j * seg->coeff_size;
+ const ftype *blockout = (const ftype *)seg->blockout->extended_data[ch] + i * seg->block_size;
+ const ctype *coeff = (const ctype *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
+
+#if DEPTH == 32
+ s->afirdsp.fcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size);
+#else
+ s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, seg->part_size);
+#endif
+
+ if (j == 0)
+ j = seg->nb_partitions;
+ j--;
+ }
+
+ seg->itx_fn(seg->itx[ch], sumout, sumin, sizeof(ftype));
+
+ buf = (ftype *)seg->buffer->extended_data[ch];
+ fn(fir_fadd)(s, buf, sumout, seg->part_size);
+
+ memcpy(dst, buf, seg->part_size * sizeof(*dst));
+
+ buf = (ftype *)seg->buffer->extended_data[ch];
+ memcpy(buf, sumout + seg->part_size, seg->part_size * sizeof(*buf));
+
+ seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ fn(fir_fadd)(s, ptr, dst, nb_samples);
+ }
+
+ if (s->min_part_size >= 8) {
+#if DEPTH == 32
+ s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
+#else
+ s->fdsp->vector_dmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 8));
+#endif
+ emms_c();
+ } else {
+ for (n = 0; n < nb_samples; n++)
+ ptr[n] *= s->wet_gain;
+ }
+
+ return 0;
+}
+
+
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