avfilter: add arbitrary audio FIR filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

configure

@@ -3083,6 +3083,8 @@ unix_protocol_select="network"
 # filters
 afftfilt_filter_deps="avcodec"
 afftfilt_filter_select="fft"
+afir_filter_deps="avcodec"
+afir_filter_select="fft"
 amovie_filter_deps="avcodec avformat"
 aresample_filter_deps="swresample"
 ass_filter_deps="libass"
@@ -6476,6 +6478,7 @@ enabled zlib && add_cppflags -DZLIB_CONST
 
 # conditional library dependencies, in linking order
 enabled afftfilt_filter     && prepend avfilter_deps "avcodec"
+enabled afir_filter         && prepend avfilter_deps "avcodec"
 enabled amovie_filter       && prepend avfilter_deps "avformat avcodec"
 enabled aresample_filter    && prepend avfilter_deps "swresample"
 enabled atempo_filter       && prepend avfilter_deps "avcodec"

doc/filters.texi

@@ -878,6 +878,49 @@ afftfilt="1-clip((b/nb)*b,0,1)"
 @end example
 @end itemize
 
+@section afir
+
+Apply an arbitrary Frequency Impulse Response filter.
+
+This filter is designed for applying long FIR filters,
+up to 30 seconds long.
+
+It can be used as component for digital crossover filters,
+room equalization, cross talk cancellation, wavefield synthesis,
+auralization, ambiophonics and ambisonics.
+
+This filter uses second stream as FIR coefficients.
+If second stream holds single channel, it will be used
+for all input channels in first stream, otherwise
+number of channels in second stream must be same as
+number of channels in first stream.
+
+It accepts the following parameters:
+
+@table @option
+@item dry
+Set dry gain. This sets input gain.
+
+@item wet
+Set wet gain. This sets final output gain.
+
+@item length
+Set Impulse Response filter length. Default is 1, which means whole IR is processed.
+
+@item again
+Enable applying gain measured from power of IR.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:
+@example
+ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
+@end example
+@end itemize
+
 @anchor{aformat}
 @section aformat
 

libavfilter/Makefile

@@ -37,6 +37,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o
 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o
 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
 OBJS-$(CONFIG_AFFTFILT_FILTER)               += af_afftfilt.o window_func.o
+OBJS-$(CONFIG_AFIR_FILTER)                   += af_afir.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
 OBJS-$(CONFIG_AGATE_FILTER)                  += af_agate.o
 OBJS-$(CONFIG_AINTERLEAVE_FILTER)            += f_interleave.o

libavfilter/af_afir.c

@@ -0,0 +1,535 @@
+/*
+ * 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
+ */
+
+/**
+ * @file
+ * An arbitrary audio FIR filter
+ */
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+#include "af_afir.h"
+
+static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
+{
+    int n;
+
+    for (n = 0; n < len; n++) {
+        const float cre = c[2 * n    ];
+        const float cim = c[2 * n + 1];
+        const float tre = t[2 * n    ];
+        const float 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 int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+    AudioFIRContext *s = ctx->priv;
+    const float *src = (const float *)s->in[0]->extended_data[ch];
+    int index1 = (s->index + 1) % 3;
+    int index2 = (s->index + 2) % 3;
+    float *sum = s->sum[ch];
+    AVFrame *out = arg;
+    float *block;
+    float *dst;
+    int n, i, j;
+
+    memset(sum, 0, sizeof(*sum) * s->fft_length);
+    block = s->block[ch] + s->part_index * s->block_size;
+    memset(block, 0, sizeof(*block) * s->fft_length);
+
+    s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, s->nb_samples);
+    emms_c();
+
+    av_rdft_calc(s->rdft[ch], block);
+    block[2 * s->part_size] = block[1];
+    block[1] = 0;
+
+    j = s->part_index;
+
+    for (i = 0; i < s->nb_partitions; i++) {
+        const int coffset = i * s->coeff_size;
+        const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
+
+        block = s->block[ch] + j * s->block_size;
+        s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
+
+        if (j == 0)
+            j = s->nb_partitions;
+        j--;
+    }
+
+    sum[1] = sum[2 * s->part_size];
+    av_rdft_calc(s->irdft[ch], sum);
+
+    dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
+    for (n = 0; n < s->part_size; n++) {
+        dst[n] += sum[n];
+    }
+
+    dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
+
+    memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
+
+    dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
+
+    if (out) {
+        float *ptr = (float *)out->extended_data[ch];
+        s->fdsp->vector_fmul_scalar(ptr, dst, s->gain * s->wet_gain, out->nb_samples);
+        emms_c();
+    }
+
+    return 0;
+}
+
+static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AVFrame *out = NULL;
+    int ret;
+
+    s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
+
+    if (!s->want_skip) {
+        out = ff_get_audio_buffer(outlink, s->nb_samples);
+        if (!out)
+            return AVERROR(ENOMEM);
+    }
+
+    s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
+    if (!s->in[0]) {
+        av_frame_free(&out);
+        return AVERROR(ENOMEM);
+    }
+
+    av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
+
+    ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
+
+    s->part_index = (s->part_index + 1) % s->nb_partitions;
+
+    av_audio_fifo_drain(s->fifo[0], s->nb_samples);
+
+    if (!s->want_skip) {
+        out->pts = s->pts;
+        if (s->pts != AV_NOPTS_VALUE)
+            s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+    }
+
+    s->index++;
+    if (s->index == 3)
+        s->index = 0;
+
+    av_frame_free(&s->in[0]);
+
+    if (s->want_skip == 1) {
+        s->want_skip = 0;
+        ret = 0;
+    } else {
+        ret = ff_filter_frame(outlink, out);
+    }
+
+    return ret;
+}
+
+static int convert_coeffs(AVFilterContext *ctx)
+{
+    AudioFIRContext *s = ctx->priv;
+    int i, ch, n, N;
+    float power = 0;
+
+    s->nb_taps = av_audio_fifo_size(s->fifo[1]);
+    if (s->nb_taps <= 0)
+        return AVERROR(EINVAL);
+
+    for (n = 4; (1 << n) < s->nb_taps; n++);
+    N = FFMIN(n, 16);
+    s->ir_length = 1 << n;
+    s->fft_length = (1 << (N + 1)) + 1;
+    s->part_size = 1 << (N - 1);
+    s->block_size = FFALIGN(s->fft_length, 32);
+    s->coeff_size = FFALIGN(s->part_size + 1, 32);
+    s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
+    s->nb_coeffs = s->ir_length + s->nb_partitions;
+
+    for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
+        s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
+        if (!s->sum[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
+        s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
+        if (!s->coeff[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
+        s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
+        if (!s->block[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
+        s->rdft[ch]  = av_rdft_init(N, DFT_R2C);
+        s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
+        if (!s->rdft[ch] || !s->irdft[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
+    if (!s->in[1])
+        return AVERROR(ENOMEM);
+
+    s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
+    if (!s->buffer)
+        return AVERROR(ENOMEM);
+
+    av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
+
+    for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
+        float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+        float *block = s->block[ch];
+        FFTComplex *coeff = s->coeff[ch];
+
+        power += s->fdsp->scalarproduct_float(time, time, s->nb_taps);
+
+        for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
+            time[i] = 0;
+
+        for (i = 0; i < s->nb_partitions; i++) {
+            const float scale = 1.f / s->part_size;
+            const int toffset = i * s->part_size;
+            const int coffset = i * s->coeff_size;
+            const int boffset = s->part_size;
+            const int remaining = s->nb_taps - (i * s->part_size);
+            const int size = remaining >= s->part_size ? s->part_size : remaining;
+
+            memset(block, 0, sizeof(*block) * s->fft_length);
+            memcpy(block + boffset, time + toffset, size * sizeof(*block));
+
+            av_rdft_calc(s->rdft[0], block);
+
+            coeff[coffset].re = block[0] * scale;
+            coeff[coffset].im = 0;
+            for (n = 1; n < s->part_size; n++) {
+                coeff[coffset + n].re = block[2 * n] * scale;
+                coeff[coffset + n].im = block[2 * n + 1] * scale;
+            }
+            coeff[coffset + s->part_size].re = block[1] * scale;
+            coeff[coffset + s->part_size].im = 0;
+        }
+    }
+
+    av_frame_free(&s->in[1]);
+    s->gain = s->again ? 1.f / sqrtf(power / ctx->inputs[1]->channels) : 1.f;
+    av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
+    av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
+    av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
+    av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
+
+    s->have_coeffs = 1;
+
+    return 0;
+}
+
+static int read_ir(AVFilterLink *link, AVFrame *frame)
+{
+    AVFilterContext *ctx = link->dst;
+    AudioFIRContext *s = ctx->priv;
+    int nb_taps, max_nb_taps;
+
+    av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
+                        frame->nb_samples);
+    av_frame_free(&frame);
+
+    nb_taps = av_audio_fifo_size(s->fifo[1]);
+    max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
+    if (nb_taps > max_nb_taps) {
+        av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
+        return AVERROR(EINVAL);
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *link, AVFrame *frame)
+{
+    AVFilterContext *ctx = link->dst;
+    AudioFIRContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    int ret = 0;
+
+    av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
+                        frame->nb_samples);
+    if (s->pts == AV_NOPTS_VALUE)
+        s->pts = frame->pts;
+
+    av_frame_free(&frame);
+
+    if (!s->have_coeffs && s->eof_coeffs) {
+        ret = convert_coeffs(ctx);
+        if (ret < 0)
+            return ret;
+    }
+
+    if (s->have_coeffs) {
+        while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) {
+            ret = fir_frame(s, outlink);
+            if (ret < 0)
+                break;
+        }
+    }
+    return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioFIRContext *s = ctx->priv;
+    int ret;
+
+    if (!s->eof_coeffs) {
+        ret = ff_request_frame(ctx->inputs[1]);
+        if (ret == AVERROR_EOF) {
+            s->eof_coeffs = 1;
+            ret = 0;
+        }
+        return ret;
+    }
+    ret = ff_request_frame(ctx->inputs[0]);
+    if (ret == AVERROR_EOF && s->have_coeffs) {
+        if (s->need_padding) {
+            AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
+
+            if (!silence)
+                return AVERROR(ENOMEM);
+            av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
+                        silence->nb_samples);
+            av_frame_free(&silence);
+            s->need_padding = 0;
+        }
+
+        while (av_audio_fifo_size(s->fifo[0]) > 0) {
+            ret = fir_frame(s, outlink);
+            if (ret < 0)
+                return ret;
+        }
+        ret = AVERROR_EOF;
+    }
+    return ret;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret, i;
+
+    layouts = ff_all_channel_counts();
+    if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
+        return ret;
+
+    for (i = 0; i < 2; i++) {
+        layouts = ff_all_channel_counts();
+        if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
+            return ret;
+    }
+
+    formats = ff_make_format_list(sample_fmts);
+    if ((ret = ff_set_common_formats(ctx, formats)) < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioFIRContext *s = ctx->priv;
+
+    if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
+        ctx->inputs[1]->channels != 1) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Second input must have same number of channels as first input or "
+               "exactly 1 channel.\n");
+        return AVERROR(EINVAL);
+    }
+
+    s->one2many = ctx->inputs[1]->channels == 1;
+    outlink->sample_rate = ctx->inputs[0]->sample_rate;
+    outlink->time_base   = ctx->inputs[0]->time_base;
+    outlink->channel_layout = ctx->inputs[0]->channel_layout;
+    outlink->channels = ctx->inputs[0]->channels;
+
+    s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
+    s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
+    if (!s->fifo[0] || !s->fifo[1])
+        return AVERROR(ENOMEM);
+
+    s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
+    s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
+    s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
+    s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
+    s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
+    if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
+        return AVERROR(ENOMEM);
+
+    s->nb_channels = outlink->channels;
+    s->nb_coef_channels = ctx->inputs[1]->channels;
+    s->want_skip = 1;
+    s->need_padding = 1;
+    s->pts = AV_NOPTS_VALUE;
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioFIRContext *s = ctx->priv;
+    int ch;
+
+    if (s->sum) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            av_freep(&s->sum[ch]);
+        }
+    }
+    av_freep(&s->sum);
+
+    if (s->coeff) {
+        for (ch = 0; ch < s->nb_coef_channels; ch++) {
+            av_freep(&s->coeff[ch]);
+        }
+    }
+    av_freep(&s->coeff);
+
+    if (s->block) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            av_freep(&s->block[ch]);
+        }
+    }
+    av_freep(&s->block);
+
+    if (s->rdft) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            av_rdft_end(s->rdft[ch]);
+        }
+    }
+    av_freep(&s->rdft);
+
+    if (s->irdft) {
+        for (ch = 0; ch < s->nb_channels; ch++) {
+            av_rdft_end(s->irdft[ch]);
+        }
+    }
+    av_freep(&s->irdft);
+
+    av_frame_free(&s->in[0]);
+    av_frame_free(&s->in[1]);
+    av_frame_free(&s->buffer);
+
+    av_audio_fifo_free(s->fifo[0]);
+    av_audio_fifo_free(s->fifo[1]);
+
+    av_freep(&s->fdsp);
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AudioFIRContext *s = ctx->priv;
+
+    s->fcmul_add = fcmul_add_c;
+
+    s->fdsp = avpriv_float_dsp_alloc(0);
+    if (!s->fdsp)
+        return AVERROR(ENOMEM);
+
+    if (ARCH_X86)
+        ff_afir_init_x86(s);
+
+    return 0;
+}
+
+static const AVFilterPad afir_inputs[] = {
+    {
+        .name           = "main",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .filter_frame   = filter_frame,
+    },{
+        .name           = "ir",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .filter_frame   = read_ir,
+    },
+    { NULL }
+};
+
+static const AVFilterPad afir_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .config_props  = config_output,
+        .request_frame = request_frame,
+    },
+    { NULL }
+};
+
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define OFFSET(x) offsetof(AudioFIRContext, x)
+
+static const AVOption afir_options[] = {
+    { "dry",    "set dry gain",     OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
+    { "wet",    "set wet gain",     OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
+    { "length", "set IR length",    OFFSET(length),   AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
+    { "again",  "enable auto gain", OFFSET(again),    AV_OPT_TYPE_BOOL,  {.i64=1}, 0, 1, AF },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(afir);
+
+AVFilter ff_af_afir = {
+    .name          = "afir",
+    .description   = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
+    .priv_size     = sizeof(AudioFIRContext),
+    .priv_class    = &afir_class,
+    .query_formats = query_formats,
+    .init          = init,
+    .uninit        = uninit,
+    .inputs        = afir_inputs,
+    .outputs       = afir_outputs,
+    .flags         = AVFILTER_FLAG_SLICE_THREADS,
+};

libavfilter/af_afir.h

@@ -0,0 +1,83 @@
+/*
+ * 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/audio_fifo.h"
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+#define MAX_IR_DURATION 30
+
+typedef struct AudioFIRContext {
+    const AVClass *class;
+
+    float wet_gain;
+    float dry_gain;
+    float length;
+    int again;
+
+    float gain;
+
+    int eof_coeffs;
+    int have_coeffs;
+    int nb_coeffs;
+    int nb_taps;
+    int part_size;
+    int part_index;
+    int coeff_size;
+    int block_size;
+    int nb_partitions;
+    int nb_channels;
+    int ir_length;
+    int fft_length;
+    int nb_coef_channels;
+    int one2many;
+    int nb_samples;
+    int want_skip;
+    int need_padding;
+
+    RDFTContext **rdft, **irdft;
+    float **sum;
+    float **block;
+    FFTComplex **coeff;
+
+    AVAudioFifo *fifo[2];
+    AVFrame *in[2];
+    AVFrame *buffer;
+    int64_t pts;
+    int index;
+
+    AVFloatDSPContext *fdsp;
+    void (*fcmul_add)(float *sum, const float *t, const float *c,
+                      ptrdiff_t len);
+} AudioFIRContext;
+
+void ff_afir_init_x86(AudioFIRContext *s);
+
+#endif /* AVFILTER_AFIR_H */

libavfilter/allfilters.c

@@ -50,6 +50,7 @@ static void register_all(void)
     REGISTER_FILTER(AEVAL,          aeval,          af);
     REGISTER_FILTER(AFADE,          afade,          af);
     REGISTER_FILTER(AFFTFILT,       afftfilt,       af);
+    REGISTER_FILTER(AFIR,           afir,           af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
     REGISTER_FILTER(AGATE,          agate,          af);
     REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  88
+#define LIBAVFILTER_VERSION_MINOR  89
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

libavfilter/x86/Makefile

@@ -1,3 +1,4 @@
+OBJS-$(CONFIG_AFIR_FILTER)                   += x86/af_afir_init.o
 OBJS-$(CONFIG_BLEND_FILTER)                  += x86/vf_blend_init.o
 OBJS-$(CONFIG_BWDIF_FILTER)                  += x86/vf_bwdif_init.o
 OBJS-$(CONFIG_COLORSPACE_FILTER)             += x86/colorspacedsp_init.o
@@ -23,6 +24,7 @@ OBJS-$(CONFIG_VOLUME_FILTER)                 += x86/af_volume_init.o
 OBJS-$(CONFIG_W3FDIF_FILTER)                 += x86/vf_w3fdif_init.o
 OBJS-$(CONFIG_YADIF_FILTER)                  += x86/vf_yadif_init.o
 
+YASM-OBJS-$(CONFIG_AFIR_FILTER)              += x86/af_afir.o
 YASM-OBJS-$(CONFIG_BLEND_FILTER)             += x86/vf_blend.o
 YASM-OBJS-$(CONFIG_BWDIF_FILTER)             += x86/vf_bwdif.o
 YASM-OBJS-$(CONFIG_COLORSPACE_FILTER)        += x86/colorspacedsp.o

libavfilter/x86/af_afir.asm

@@ -0,0 +1,60 @@
+;*****************************************************************************
+;* x86-optimized functions for afir filter
+;* 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 "libavutil/x86/x86util.asm"
+
+SECTION .text
+
+;------------------------------------------------------------------------------
+; void ff_fcmul_add(float *sum, const float *t, const float *c, int len)
+;------------------------------------------------------------------------------
+
+INIT_XMM sse3
+cglobal fcmul_add, 4,4,6, sum, t, c, len
+    shl       lend, 3
+    add       lend, mmsize*2
+    add         tq, lenq
+    add         cq, lenq
+    add       sumq, lenq
+    neg       lenq
+ALIGN 16
+.loop:
+    movsldup  m0, [tq + lenq]
+    movsldup  m3, [tq + lenq+mmsize]
+    movaps    m1, [cq + lenq]
+    movaps    m4, [cq + lenq+mmsize]
+    mulps     m0, m1
+    mulps     m3, m4
+    shufps    m1, m1, 0xb1
+    shufps    m4, m4, 0xb1
+    movshdup  m2, [tq + lenq]
+    movshdup  m5, [tq + lenq+mmsize]
+    mulps     m2, m1
+    mulps     m5, m4
+    addsubps  m0, m2
+    addsubps  m3, m5
+    addps     m0, [sumq + lenq]
+    addps     m3, [sumq + lenq+mmsize]
+    movaps    [sumq + lenq], m0
+    movaps    [sumq + lenq+mmsize], m3
+    add       lenq, mmsize*2
+    jl .loop
+    REP_RET

libavfilter/x86/af_afir_init.c

@@ -0,0 +1,35 @@
+/*
+ * 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 "config.h"
+#include "libavutil/attributes.h"
+#include "libavutil/cpu.h"
+#include "libavutil/x86/cpu.h"
+#include "libavfilter/af_afir.h"
+
+void ff_fcmul_add_sse3(float *sum, const float *t, const float *c,
+                       ptrdiff_t len);
+
+av_cold void ff_afir_init_x86(AudioFIRContext *s)
+{
+    int cpu_flags = av_get_cpu_flags();
+
+    if (EXTERNAL_SSE3(cpu_flags)) {
+        s->fcmul_add = ff_fcmul_add_sse3;
+    }
+}