avfilter/af_afir: add support for double sample format

1 files changed, 392 insertions, 0 deletions

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;
+}
+
+