avfilter/af_afir: add support for double sample format

4 files changed, 607 insertions, 415 deletions

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;
+}
+
+