avfilter: add afftfilter

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

7 files changed, 485 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 59bfbb1b72..4e37f00c53 100644
--- a/Changelog
+++ b/Changelog
@@ -56,6 +56,7 @@ version <next>:
 - ahistogram filter
 - only seek with the right mouse button in ffplay
 - toggle full screen when double-clicking with the left mouse button in ffplay
+- afftfilt filter
 
 
 version 2.8:
diff --git a/configure b/configure
index 04d83ebcca..ef5a71289c 100755
--- a/configure
+++ b/configure
@@ -2841,6 +2841,8 @@ unix_protocol_deps="sys_un_h"
 unix_protocol_select="network"
 
 # filters
+afftfilt_filter_deps="avcodec"
+afftfilt_filter_select="fft"
 amovie_filter_deps="avcodec avformat"
 aresample_filter_deps="swresample"
 ass_filter_deps="libass"
@@ -6065,6 +6067,7 @@ done
 enabled zlib && add_cppflags -DZLIB_CONST
 
 # conditional library dependencies, in linking order
+enabled afftfilt_filter     && prepend avfilter_deps "avcodec"
 enabled amovie_filter       && prepend avfilter_deps "avformat avcodec"
 enabled aresample_filter    && prepend avfilter_deps "swresample"
 enabled asyncts_filter      && prepend avfilter_deps "avresample"
diff --git a/doc/filters.texi b/doc/filters.texi
index f5f4bfc3c5..053e229a5e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -733,6 +733,83 @@ afade=t=out:st=875:d=25
 @end example
 @end itemize
 
+@section afftfilt
+Apply arbitrary expressions to samples in frequency domain.
+
+@table @option
+@item real
+Set frequency domain real expression for each separate channel separated
+by '|'. Default is "1".
+If the number of input channels is greater than the number of
+expressions, the last specified expression is used for the remaining
+output channels.
+
+@item imag
+Set frequency domain imaginary expression for each separate channel
+separated by '|'. If not set, @var{real} option is used.
+
+Each expression in @var{real} and @var{imag} can contain the following
+constants:
+
+@table @option
+@item sr
+sample rate
+
+@item b
+current frequency bin number
+
+@item nb
+number of available bins
+
+@item ch
+channel number of the current expression
+
+@item chs
+number of channels
+
+@item pts
+current frame pts
+@end table
+
+@item win_size
+Set window size.
+
+It accepts the following values:
+@table @samp
+@item w16
+@item w32
+@item w64
+@item w128
+@item w256
+@item w512
+@item w1024
+@item w2048
+@item w4096
+@item w8192
+@item w16384
+@item w32768
+@item w65536
+@end table
+Default is @code{w4096}
+
+@item win_func
+Set window function. Default is @code{hann}.
+
+@item overlap
+Set window overlap. If set to 1, the recommended overlap for selected
+window function will be picked. Default is @code{0.75}.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Increase first 50 bins by 0.1 and lower all other frequencies by factor of 10:
+@example
+afftfilt="1.1*between(b\,0\,49)+0.1*between(b\,50\,f)"
+@end example
+@end itemize
+
 @anchor{aformat}
 @section aformat
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e3e3561933..242f56dbf3 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -29,6 +29,7 @@ OBJS-$(CONFIG_ACROSSFADE_FILTER)             += af_afade.o
 OBJS-$(CONFIG_ADELAY_FILTER)                 += af_adelay.o
 OBJS-$(CONFIG_AECHO_FILTER)                  += af_aecho.o
 OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o
+OBJS-$(CONFIG_AFFTFILT_FILTER)               += af_afftfilt.o window_func.o
 OBJS-$(CONFIG_ANEQUALIZER_FILTER)            += af_anequalizer.o
 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o
 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
diff --git a/libavfilter/af_afftfilt.c b/libavfilter/af_afftfilt.c
new file mode 100644
index 0000000000..8c75b4f0ae
--- /dev/null
+++ b/libavfilter/af_afftfilt.c
@@ -0,0 +1,401 @@
+/*
+ * Copyright (c) 2016 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/audio_fifo.h"
+#include "libavutil/avstring.h"
+#include "libavfilter/internal.h"
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+#include "libavutil/eval.h"
+#include "audio.h"
+#include "window_func.h"
+
+typedef struct AFFTFiltContext {
+    const AVClass *class;
+    char *real_str;
+    char *img_str;
+    int fft_bits;
+
+    FFTContext *fft, *ifft;
+    FFTComplex **fft_data;
+    int nb_exprs;
+    int window_size;
+    AVExpr **real;
+    AVExpr **imag;
+    AVAudioFifo *fifo;
+    int64_t pts;
+    int hop_size;
+    float overlap;
+    AVFrame *buffer;
+    int start, end;
+    int win_func;
+    float win_scale;
+    float *window_func_lut;
+} AFFTFiltContext;
+
+static const char *const var_names[] = {            "sr",     "b",       "nb",        "ch",        "chs",   "pts",        NULL };
+enum                                   { VAR_SAMPLE_RATE, VAR_BIN, VAR_NBBINS, VAR_CHANNEL, VAR_CHANNELS, VAR_PTS, VAR_VARS_NB };
+
+#define OFFSET(x) offsetof(AFFTFiltContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption afftfilt_options[] = {
+    { "real", "set channels real expressions",       OFFSET(real_str), AV_OPT_TYPE_STRING, {.str = "1" }, 0, 0, A },
+    { "imag",  "set channels imaginary expressions", OFFSET(img_str),  AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, A },
+    { "win_size", "set window size", OFFSET(fft_bits), AV_OPT_TYPE_INT, {.i64=12}, 4, 16, A, "fft" },
+        { "w16",    0, 0, AV_OPT_TYPE_CONST, {.i64=4},  0, 0, A, "fft" },
+        { "w32",    0, 0, AV_OPT_TYPE_CONST, {.i64=5},  0, 0, A, "fft" },
+        { "w64",    0, 0, AV_OPT_TYPE_CONST, {.i64=6},  0, 0, A, "fft" },
+        { "w128",   0, 0, AV_OPT_TYPE_CONST, {.i64=7},  0, 0, A, "fft" },
+        { "w256",   0, 0, AV_OPT_TYPE_CONST, {.i64=8},  0, 0, A, "fft" },
+        { "w512",   0, 0, AV_OPT_TYPE_CONST, {.i64=9},  0, 0, A, "fft" },
+        { "w1024",  0, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, A, "fft" },
+        { "w2048",  0, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, A, "fft" },
+        { "w4096",  0, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, A, "fft" },
+        { "w8192",  0, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, A, "fft" },
+        { "w16384", 0, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, A, "fft" },
+        { "w32768", 0, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, A, "fft" },
+        { "w65536", 0, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, A, "fft" },
+    { "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64 = WFUNC_HANNING}, 0, NB_WFUNC-1, A, "win_func" },
+        { "rect",     "Rectangular",      0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT},     0, 0, A, "win_func" },
+        { "bartlett", "Bartlett",         0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, A, "win_func" },
+        { "hann",     "Hann",             0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING},  0, 0, A, "win_func" },
+        { "hanning",  "Hanning",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING},  0, 0, A, "win_func" },
+        { "hamming",  "Hamming",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING},  0, 0, A, "win_func" },
+        { "sine",     "Sine",             0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE},     0, 0, A, "win_func" },
+    { "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0,  1, A },
+    { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(afftfilt);
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AFFTFiltContext *s = ctx->priv;
+    char *saveptr = NULL;
+    int ret = 0, ch, i;
+    float overlap;
+    char *args, *last_expr = NULL;
+
+    s->fft  = av_fft_init(s->fft_bits, 0);
+    s->ifft = av_fft_init(s->fft_bits, 1);
+    if (!s->fft || !s->ifft)
+        return AVERROR(ENOMEM);
+
+    s->window_size = 1 << s->fft_bits;
+
+    s->fft_data = av_calloc(inlink->channels, sizeof(*s->fft_data));
+    if (!s->fft_data)
+        return AVERROR(ENOMEM);
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        s->fft_data[ch] = av_calloc(s->window_size, sizeof(**s->fft_data));
+        if (!s->fft_data[ch])
+            return AVERROR(ENOMEM);
+    }
+
+    s->real = av_calloc(inlink->channels, sizeof(*s->real));
+    if (!s->real)
+        return AVERROR(ENOMEM);
+
+    s->imag = av_calloc(inlink->channels, sizeof(*s->imag));
+    if (!s->imag)
+        return AVERROR(ENOMEM);
+
+    args = av_strdup(s->real_str);
+    if (!args)
+        return AVERROR(ENOMEM);
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        char *arg = av_strtok(ch == 0 ? args : NULL, "|", &saveptr);
+
+        ret = av_expr_parse(&s->real[ch], arg ? arg : last_expr, var_names,
+                            NULL, NULL, NULL, NULL, 0, ctx);
+        if (ret < 0)
+            break;
+        if (arg)
+            last_expr = arg;
+        s->nb_exprs++;
+    }
+
+    av_free(args);
+
+    args = av_strdup(s->img_str ? s->img_str : s->real_str);
+    if (!args)
+        return AVERROR(ENOMEM);
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        char *arg = av_strtok(ch == 0 ? args : NULL, "|", &saveptr);
+
+        ret = av_expr_parse(&s->imag[ch], arg ? arg : last_expr, var_names,
+                            NULL, NULL, NULL, NULL, 0, ctx);
+        if (ret < 0)
+            break;
+        if (arg)
+            last_expr = arg;
+    }
+
+    av_free(args);
+
+    s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+
+    s->window_func_lut = av_realloc_f(s->window_func_lut, s->window_size,
+                                      sizeof(*s->window_func_lut));
+    if (!s->window_func_lut)
+        return AVERROR(ENOMEM);
+    ff_generate_window_func(s->window_func_lut, s->window_size, s->win_func, &overlap);
+    if (s->overlap == 1)
+        s->overlap = overlap;
+
+    for (s->win_scale = 0, i = 0; i < s->window_size; i++) {
+        s->win_scale += s->window_func_lut[i] * s->window_func_lut[i];
+    }
+
+    s->hop_size = s->window_size * (1 - s->overlap);
+    if (s->hop_size <= 0)
+        return AVERROR(EINVAL);
+
+    s->buffer = ff_get_audio_buffer(inlink, s->window_size * 2);
+    if (!s->buffer)
+        return AVERROR(ENOMEM);
+
+    return ret;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AFFTFiltContext *s = ctx->priv;
+    const int window_size = s->window_size;
+    const float f = 1. / s->win_scale;
+    double values[VAR_VARS_NB];
+    AVFrame *out, *in = NULL;
+    int ch, n, ret, i, j, k;
+    int start = s->start, end = s->end;
+
+    av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
+    av_frame_free(&frame);
+
+    while (av_audio_fifo_size(s->fifo) >= window_size) {
+        if (!in) {
+            in = ff_get_audio_buffer(outlink, window_size);
+            if (!in)
+                return AVERROR(ENOMEM);
+        }
+
+        ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, window_size);
+        if (ret < 0)
+            break;
+
+        for (ch = 0; ch < inlink->channels; ch++) {
+            const float *src = (float *)in->extended_data[ch];
+            FFTComplex *fft_data = s->fft_data[ch];
+
+            for (n = 0; n < in->nb_samples; n++) {
+                fft_data[n].re = src[n] * s->window_func_lut[n];
+                fft_data[n].im = 0;
+            }
+
+            for (; n < window_size; n++) {
+                fft_data[n].re = 0;
+                fft_data[n].im = 0;
+            }
+        }
+
+        values[VAR_PTS]         = s->pts;
+        values[VAR_SAMPLE_RATE] = inlink->sample_rate;
+        values[VAR_NBBINS]      = window_size / 2;
+        values[VAR_CHANNELS]    = inlink->channels;
+
+        for (ch = 0; ch < inlink->channels; ch++) {
+            FFTComplex *fft_data = s->fft_data[ch];
+            float *buf = (float *)s->buffer->extended_data[ch];
+            int x;
+
+            values[VAR_CHANNEL] = ch;
+
+            av_fft_permute(s->fft, fft_data);
+            av_fft_calc(s->fft, fft_data);
+
+            for (n = 0; n < window_size / 2; n++) {
+                float fr, fi;
+
+                values[VAR_BIN] = n;
+
+                fr = av_expr_eval(s->real[ch], values, s);
+                fi = av_expr_eval(s->imag[ch], values, s);
+
+                fft_data[n].re *= fr;
+                fft_data[n].im *= fi;
+            }
+
+            for (n = window_size / 2 + 1, x = window_size / 2 - 1; n < window_size; n++, x--) {
+                fft_data[n].re =  fft_data[x].re;
+                fft_data[n].im = -fft_data[x].im;
+            }
+
+            av_fft_permute(s->ifft, fft_data);
+            av_fft_calc(s->ifft, fft_data);
+
+            start = s->start;
+            end = s->end;
+            k = end;
+            for (i = 0, j = start; j < k && i < window_size; i++, j++) {
+                buf[j] += s->fft_data[ch][i].re * f;
+            }
+
+            for (; i < window_size; i++, j++) {
+                buf[j] = s->fft_data[ch][i].re * f;
+            }
+
+            start += s->hop_size;
+            end = j;
+        }
+
+        s->start = start;
+        s->end = end;
+
+        if (start >= window_size) {
+            float *dst, *buf;
+
+            start -= window_size;
+            end   -= window_size;
+
+            s->start = start;
+            s->end = end;
+
+            out = ff_get_audio_buffer(outlink, window_size);
+            if (!out) {
+                ret = AVERROR(ENOMEM);
+                break;
+            }
+
+            out->pts = s->pts;
+            s->pts += window_size;
+
+            for (ch = 0; ch < inlink->channels; ch++) {
+                dst = (float *)out->extended_data[ch];
+                buf = (float *)s->buffer->extended_data[ch];
+
+                for (n = 0; n < window_size; n++) {
+                    dst[n] = buf[n] * (1 - s->overlap);
+                }
+                memmove(buf, buf + window_size, window_size * 4);
+            }
+
+            ret = ff_filter_frame(outlink, out);
+            if (ret < 0)
+                break;
+        }
+
+        av_audio_fifo_drain(s->fifo, s->hop_size);
+    }
+
+    av_frame_free(&in);
+    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;
+
+    layouts = ff_all_channel_counts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_channel_layouts(ctx, layouts);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AFFTFiltContext *s = ctx->priv;
+    int i;
+
+    av_fft_end(s->fft);
+    av_fft_end(s->ifft);
+
+    for (i = 0; i < s->nb_exprs; i++) {
+        if (s->fft_data)
+            av_freep(&s->fft_data[i]);
+    }
+    av_freep(&s->fft_data);
+
+    for (i = 0; i < s->nb_exprs; i++) {
+        av_expr_free(s->real[i]);
+        av_expr_free(s->imag[i]);
+    }
+
+    av_freep(&s->real);
+    av_freep(&s->imag);
+    av_frame_free(&s->buffer);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_afftfilt = {
+    .name            = "afftfilt",
+    .description     = NULL_IF_CONFIG_SMALL("Apply arbitrary expressions to samples in frequency domain."),
+    .priv_size       = sizeof(AFFTFiltContext),
+    .priv_class      = &afftfilt_class,
+    .inputs          = inputs,
+    .outputs         = outputs,
+    .query_formats   = query_formats,
+    .uninit          = uninit,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 1faf39301a..f270bdf3a2 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -52,6 +52,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(AEMPHASIS,      aemphasis,      af);
     REGISTER_FILTER(AEVAL,          aeval,          af);
     REGISTER_FILTER(AFADE,          afade,          af);
+    REGISTER_FILTER(AFFTFILT,       afftfilt,       af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
     REGISTER_FILTER(AGATE,          agate,          af);
     REGISTER_FILTER(AINTERLEAVE,    ainterleave,    af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index d1f3802b68..e6c7aebd8f 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  25
+#define LIBAVFILTER_VERSION_MINOR  26
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \