avfilter: add showfreqs filter

11 files changed, 750 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 0e87c099b8..099f0a22c0 100644
--- a/Changelog
+++ b/Changelog
@@ -31,6 +31,7 @@ version <next>:
 - atadenoise video filter
 - OS X VideoToolbox support
 - aphasemeter filter
+- showfreqs filter
 
 
 version 2.7:
diff --git a/configure b/configure
index bc2160df7c..8d104e7ff2 100755
--- a/configure
+++ b/configure
@@ -2785,6 +2785,8 @@ select_filter_select="pixelutils"
 smartblur_filter_deps="gpl swscale"
 showcqt_filter_deps="avcodec"
 showcqt_filter_select="fft"
+showfreqs_filter_deps="avcodec"
+showfreqs_filter_select="fft"
 showspectrum_filter_deps="avcodec"
 showspectrum_filter_select="rdft"
 spp_filter_deps="gpl avcodec"
diff --git a/doc/filters.texi b/doc/filters.texi
index e94ec40fc7..360ff6aca7 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -12704,6 +12704,121 @@ gamma=2:gamma2=2
 
 @end itemize
 
+@section showfreqs
+
+Convert input audio to video output representing the audio power spectrum.
+Audio amplitude is on Y-axis while frequency is on X-axis.
+
+The filter accepts the following options:
+
+@table @option
+@item size, s
+Specify size of video. For the syntax of this option, check the
+@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}.
+Default is @code{1024x512}.
+
+@item mode
+Set display mode.
+This set how each frequency bin will be represented.
+
+It accepts the following values:
+@table @samp
+@item line
+@item bar
+@item dot
+@end table
+Default is @code{bar}.
+
+@item ascale
+Set amplitude scale.
+
+It accepts the following values:
+@table @samp
+@item lin
+Linear scale.
+
+@item sqrt
+Square root scale.
+
+@item cbrt
+Cubic root scale.
+
+@item log
+Logarithmic scale.
+@end table
+Default is @code{log}.
+
+@item fscale
+Set frequency scale.
+
+It accepts the following values:
+@table @samp
+@item lin
+Linear scale.
+
+@item log
+Logarithmic scale.
+
+@item rlog
+Reverse logarithmic scale.
+@end table
+Default is @code{lin}.
+
+@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{w2048}
+
+@item win_func
+Set windowing function.
+
+It accepts the following values:
+@table @samp
+@item rect
+@item bartlett
+@item hanning
+@item hamming
+@item blackman
+@item welch
+@item flattop
+@item bharris
+@item bnuttall
+@item bhann
+@item sine
+@item nuttall
+@end table
+Default is @code{hanning}.
+
+@item overlap
+Set window overlap. In range @code{[0, 1]}. Default is @code{1},
+which means optimal overlap for selected window function will be picked.
+
+@item averaging
+Set time averaging. Setting this to 0 will display current maximal peaks.
+Default is @code{1}, which means time averaging is disabled.
+
+@item color
+Specify list of colors separated by space or by '|' which will be used to
+draw channel frequencies. Unrecognized or missing colors will be replaced
+by white color.
+@end table
+
 @section showspectrum
 
 Convert input audio to a video output, representing the audio frequency
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 75581f2cf7..d2e944fd2a 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -258,6 +258,7 @@ OBJS-$(CONFIG_APHASEMETER_FILTER)            += avf_aphasemeter.o
 OBJS-$(CONFIG_AVECTORSCOPE_FILTER)           += avf_avectorscope.o
 OBJS-$(CONFIG_CONCAT_FILTER)                 += avf_concat.o
 OBJS-$(CONFIG_SHOWCQT_FILTER)                += avf_showcqt.o
+OBJS-$(CONFIG_SHOWFREQS_FILTER)              += avf_showfreqs.o
 OBJS-$(CONFIG_SHOWSPECTRUM_FILTER)           += avf_showspectrum.o
 OBJS-$(CONFIG_SHOWVOLUME_FILTER)             += avf_showvolume.o
 OBJS-$(CONFIG_SHOWWAVES_FILTER)              += avf_showwaves.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ce5138220d..23d0510bc9 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -273,6 +273,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(AVECTORSCOPE,   avectorscope,   avf);
     REGISTER_FILTER(CONCAT,         concat,         avf);
     REGISTER_FILTER(SHOWCQT,        showcqt,        avf);
+    REGISTER_FILTER(SHOWFREQS,      showfreqs,      avf);
     REGISTER_FILTER(SHOWSPECTRUM,   showspectrum,   avf);
     REGISTER_FILTER(SHOWVOLUME,     showvolume,     avf);
     REGISTER_FILTER(SHOWWAVES,      showwaves,      avf);
diff --git a/libavfilter/avf_showfreqs.c b/libavfilter/avf_showfreqs.c
new file mode 100644
index 0000000000..a1ef0bf816
--- /dev/null
+++ b/libavfilter/avf_showfreqs.c
@@ -0,0 +1,558 @@
+/*
+ * Copyright (c) 2015 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 <math.h>
+
+#include "libavcodec/avfft.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
+#include "libavutil/parseutils.h"
+#include "audio.h"
+#include "video.h"
+#include "avfilter.h"
+#include "internal.h"
+
+enum DisplayMode    { LINE, BAR, DOT, NB_MODES };
+enum FrequencyScale { FS_LINEAR, FS_LOG, FS_RLOG, NB_FSCALES };
+enum AmplitudeScale { AS_LINEAR, AS_SQRT, AS_CBRT, AS_LOG, NB_ASCALES };
+enum WindowFunc     { WFUNC_RECT, WFUNC_HANNING, WFUNC_HAMMING, WFUNC_BLACKMAN,
+                      WFUNC_BARTLETT, WFUNC_WELCH, WFUNC_FLATTOP,
+                      WFUNC_BHARRIS, WFUNC_BNUTTALL, WFUNC_SINE, WFUNC_NUTTALL,
+                      WFUNC_BHANN, NB_WFUNC };
+
+typedef struct ShowFreqsContext {
+    const AVClass *class;
+    int w, h;
+    int mode;
+    int fft_bits;
+    int ascale, fscale;
+    int avg;
+    int win_func;
+    FFTContext *fft;
+    FFTComplex **fft_data;
+    float **avg_data;
+    float *window_func_lut;
+    float overlap;
+    int skip_samples;
+    int nb_channels;
+    int nb_freq;
+    int win_size;
+    float scale;
+    char *colors;
+    AVAudioFifo *fifo;
+    int64_t pts;
+} ShowFreqsContext;
+
+#define OFFSET(x) offsetof(ShowFreqsContext, x)
+#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
+
+static const AVOption showfreqs_options[] = {
+    { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "1024x512"}, 0, 0, FLAGS },
+    { "s",    "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "1024x512"}, 0, 0, FLAGS },
+    { "mode", "set display mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=BAR}, 0, NB_MODES-1, FLAGS, "mode" },
+        { "line", "show lines",  0, AV_OPT_TYPE_CONST, {.i64=LINE},   0, 0, FLAGS, "mode" },
+        { "bar",  "show bars",   0, AV_OPT_TYPE_CONST, {.i64=BAR},    0, 0, FLAGS, "mode" },
+        { "dot",  "show dots",   0, AV_OPT_TYPE_CONST, {.i64=DOT},    0, 0, FLAGS, "mode" },
+    { "ascale", "set amplitude scale", OFFSET(ascale), AV_OPT_TYPE_INT, {.i64=AS_LOG}, 0, NB_ASCALES-1, FLAGS, "ascale" },
+        { "lin",  "linear",      0, AV_OPT_TYPE_CONST, {.i64=AS_LINEAR}, 0, 0, FLAGS, "ascale" },
+        { "sqrt", "square root", 0, AV_OPT_TYPE_CONST, {.i64=AS_SQRT},   0, 0, FLAGS, "ascale" },
+        { "cbrt", "cubic root",  0, AV_OPT_TYPE_CONST, {.i64=AS_CBRT},   0, 0, FLAGS, "ascale" },
+        { "log",  "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=AS_LOG},    0, 0, FLAGS, "ascale" },
+    { "fscale", "set frequency scale", OFFSET(fscale), AV_OPT_TYPE_INT, {.i64=FS_LINEAR}, 0, NB_FSCALES-1, FLAGS, "fscale" },
+        { "lin",  "linear",              0, AV_OPT_TYPE_CONST, {.i64=FS_LINEAR}, 0, 0, FLAGS, "fscale" },
+        { "log",  "logarithmic",         0, AV_OPT_TYPE_CONST, {.i64=FS_LOG},    0, 0, FLAGS, "fscale" },
+        { "rlog", "reverse logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=FS_RLOG},   0, 0, FLAGS, "fscale" },
+    { "win_size", "set window size", OFFSET(fft_bits), AV_OPT_TYPE_INT, {.i64=11}, 4, 16, FLAGS, "fft" },
+        { "w16",    0, 0, AV_OPT_TYPE_CONST, {.i64=4},  0, 0, FLAGS, "fft" },
+        { "w32",    0, 0, AV_OPT_TYPE_CONST, {.i64=5},  0, 0, FLAGS, "fft" },
+        { "w64",    0, 0, AV_OPT_TYPE_CONST, {.i64=6},  0, 0, FLAGS, "fft" },
+        { "w128",   0, 0, AV_OPT_TYPE_CONST, {.i64=7},  0, 0, FLAGS, "fft" },
+        { "w256",   0, 0, AV_OPT_TYPE_CONST, {.i64=8},  0, 0, FLAGS, "fft" },
+        { "w512",   0, 0, AV_OPT_TYPE_CONST, {.i64=9},  0, 0, FLAGS, "fft" },
+        { "w1024",  0, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, FLAGS, "fft" },
+        { "w2048",  0, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, FLAGS, "fft" },
+        { "w4096",  0, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, FLAGS, "fft" },
+        { "w8192",  0, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, FLAGS, "fft" },
+        { "w16384", 0, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, FLAGS, "fft" },
+        { "w32768", 0, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, FLAGS, "fft" },
+        { "w65536", 0, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, FLAGS, "fft" },
+    { "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64=WFUNC_HANNING}, 0, NB_WFUNC-1, FLAGS, "win_func" },
+        { "rect",     "Rectangular",      0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT},     0, 0, FLAGS, "win_func" },
+        { "bartlett", "Bartlett",         0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, FLAGS, "win_func" },
+        { "hanning",  "Hanning",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING},  0, 0, FLAGS, "win_func" },
+        { "hamming",  "Hamming",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING},  0, 0, FLAGS, "win_func" },
+        { "blackman", "Blackman",         0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BLACKMAN}, 0, 0, FLAGS, "win_func" },
+        { "welch",    "Welch",            0, AV_OPT_TYPE_CONST, {.i64=WFUNC_WELCH},    0, 0, FLAGS, "win_func" },
+        { "flattop",  "Flat-top",         0, AV_OPT_TYPE_CONST, {.i64=WFUNC_FLATTOP},  0, 0, FLAGS, "win_func" },
+        { "bharris",  "Blackman-Harris",  0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHARRIS},  0, 0, FLAGS, "win_func" },
+        { "bnuttall", "Blackman-Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BNUTTALL}, 0, 0, FLAGS, "win_func" },
+        { "bhann",    "Bartlett-Hann",    0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHANN},    0, 0, FLAGS, "win_func" },
+        { "sine",     "Sine",             0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE},     0, 0, FLAGS, "win_func" },
+        { "nuttall",  "Nuttall",          0, AV_OPT_TYPE_CONST, {.i64=WFUNC_NUTTALL},  0, 0, FLAGS, "win_func" },
+    { "overlap",  "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=1.}, 0., 1., FLAGS },
+    { "averaging", "set time averaging", OFFSET(avg), AV_OPT_TYPE_INT, {.i64=1}, 0, INT32_MAX, FLAGS },
+    { "colors", "set channels colors", OFFSET(colors), AV_OPT_TYPE_STRING, {.str = "red|green|blue|yellow|orange|lime|pink|magenta|brown" }, 0, 0, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(showfreqs);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    AVFilterLink *inlink = ctx->inputs[0];
+    AVFilterLink *outlink = ctx->outputs[0];
+    static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
+    static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };
+
+    /* set input audio formats */
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_formats_ref(formats, &inlink->out_formats);
+
+    layouts = ff_all_channel_layouts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts);
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_formats_ref(formats, &inlink->out_samplerates);
+
+    /* set output video format */
+    formats = ff_make_format_list(pix_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_formats_ref(formats, &outlink->in_formats);
+
+    return 0;
+}
+
+static void generate_window_func(float *lut, int N, int win_func, float *overlap)
+{
+    int n;
+
+    switch (win_func) {
+    case WFUNC_RECT:
+        for (n = 0; n < N; n++)
+            lut[n] = 1.;
+        *overlap = 0.;
+        break;
+    case WFUNC_BARTLETT:
+        for (n = 0; n < N; n++)
+            lut[n] = 1.-FFABS((n-(N-1)/2.)/((N-1)/2.));
+        *overlap = 0.5;
+        break;
+    case WFUNC_HANNING:
+        for (n = 0; n < N; n++)
+            lut[n] = .5*(1-cos(2*M_PI*n/(N-1)));
+        *overlap = 0.5;
+        break;
+    case WFUNC_HAMMING:
+        for (n = 0; n < N; n++)
+            lut[n] = .54-.46*cos(2*M_PI*n/(N-1));
+        *overlap = 0.5;
+        break;
+    case WFUNC_BLACKMAN:
+        for (n = 0; n < N; n++)
+            lut[n] = .42659-.49656*cos(2*M_PI*n/(N-1))+.076849*cos(4*M_PI*n/(N-1));
+        *overlap = 0.661;
+        break;
+    case WFUNC_WELCH:
+        for (n = 0; n < N; n++)
+            lut[n] = 1.-(n-(N-1)/2.)/((N-1)/2.)*(n-(N-1)/2.)/((N-1)/2.);
+        *overlap = 0.293;
+        break;
+    case WFUNC_FLATTOP:
+        for (n = 0; n < N; n++)
+            lut[n] = 1.-1.985844164102*cos( 2*M_PI*n/(N-1))+1.791176438506*cos( 4*M_PI*n/(N-1))-
+                        1.282075284005*cos( 6*M_PI*n/(N-1))+0.667777530266*cos( 8*M_PI*n/(N-1))-
+                        0.240160796576*cos(10*M_PI*n/(N-1))+0.056656381764*cos(12*M_PI*n/(N-1))-
+                        0.008134974479*cos(14*M_PI*n/(N-1))+0.000624544650*cos(16*M_PI*n/(N-1))-
+                        0.000019808998*cos(18*M_PI*n/(N-1))+0.000000132974*cos(20*M_PI*n/(N-1));
+        *overlap = 0.841;
+        break;
+    case WFUNC_BHARRIS:
+        for (n = 0; n < N; n++)
+            lut[n] = 0.35875-0.48829*cos(2*M_PI*n/(N-1))+0.14128*cos(4*M_PI*n/(N-1))-0.01168*cos(6*M_PI*n/(N-1));
+        *overlap = 0.661;
+        break;
+    case WFUNC_BNUTTALL:
+        for (n = 0; n < N; n++)
+            lut[n] = 0.3635819-0.4891775*cos(2*M_PI*n/(N-1))+0.1365995*cos(4*M_PI*n/(N-1))-0.0106411*cos(6*M_PI*n/(N-1));
+        *overlap = 0.661;
+        break;
+    case WFUNC_BHANN:
+        for (n = 0; n < N; n++)
+            lut[n] = 0.62-0.48*FFABS(n/(N-1)-.5)-0.38*cos(2*M_PI*n/(N-1));
+        *overlap = 0.5;
+        break;
+    case WFUNC_SINE:
+        for (n = 0; n < N; n++)
+            lut[n] = sin(M_PI*n/(N-1));
+        *overlap = 0.75;
+        break;
+    case WFUNC_NUTTALL:
+        for (n = 0; n < N; n++)
+            lut[n] = 0.355768-0.487396*cos(2*M_PI*n/(N-1))+0.144232*cos(4*M_PI*n/(N-1))-0.012604*cos(6*M_PI*n/(N-1));
+        *overlap = 0.663;
+        break;
+    default:
+        av_assert0(0);
+    }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AVFilterLink *inlink = ctx->inputs[0];
+    ShowFreqsContext *s = ctx->priv;
+    float overlap;
+    int i;
+
+    s->nb_freq = 1 << (s->fft_bits - 1);
+    s->win_size = s->nb_freq << 1;
+    av_audio_fifo_free(s->fifo);
+    av_fft_end(s->fft);
+    s->fft = av_fft_init(s->fft_bits, 0);
+    if (!s->fft) {
+        av_log(ctx, AV_LOG_ERROR, "Unable to create FFT context. "
+               "The window size might be too high.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* FFT buffers: x2 for each (display) channel buffer.
+     * Note: we use free and malloc instead of a realloc-like function to
+     * make sure the buffer is aligned in memory for the FFT functions. */
+    for (i = 0; i < s->nb_channels; i++) {
+        av_freep(&s->fft_data[i]);
+        av_freep(&s->avg_data[i]);
+    }
+    av_freep(&s->fft_data);
+    av_freep(&s->avg_data);
+    s->nb_channels = inlink->channels;
+
+    s->fft_data = av_calloc(s->nb_channels, sizeof(*s->fft_data));
+    if (!s->fft_data)
+        return AVERROR(ENOMEM);
+    s->avg_data = av_calloc(s->nb_channels, sizeof(*s->avg_data));
+    if (!s->fft_data)
+        return AVERROR(ENOMEM);
+    for (i = 0; i < s->nb_channels; i++) {
+        s->fft_data[i] = av_calloc(s->win_size, sizeof(**s->fft_data));
+        s->avg_data[i] = av_calloc(s->nb_freq, sizeof(**s->avg_data));
+        if (!s->fft_data[i] || !s->avg_data[i])
+            return AVERROR(ENOMEM);
+    }
+
+    /* pre-calc windowing function */
+    s->window_func_lut = av_realloc_f(s->window_func_lut, s->win_size,
+                                      sizeof(*s->window_func_lut));
+    if (!s->window_func_lut)
+        return AVERROR(ENOMEM);
+    generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap);
+    if (s->overlap == 1.)
+        s->overlap = overlap;
+    s->skip_samples = (1. - s->overlap) * s->win_size;
+    if (s->skip_samples < 1) {
+        av_log(ctx, AV_LOG_ERROR, "overlap %f too big\n", s->overlap);
+        return AVERROR(EINVAL);
+    }
+
+    for (s->scale = 0, i = 0; i < s->win_size; i++) {
+        s->scale += s->window_func_lut[i] * s->window_func_lut[i];
+    }
+
+    outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
+    outlink->frame_rate = av_make_q(inlink->sample_rate, s->win_size * (1.-s->overlap));
+    outlink->sample_aspect_ratio = (AVRational){1,1};
+    outlink->w = s->w;
+    outlink->h = s->h;
+
+    s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->win_size);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+    return 0;
+}
+
+static inline void draw_dot(AVFrame *out, int x, int y, uint8_t fg[4])
+{
+
+    uint32_t color = AV_RL32(out->data[0] + y * out->linesize[0] + x * 4);
+
+    if ((color & 0xffffff) != 0)
+        AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg) | color);
+    else
+        AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
+}
+
+static int get_sx(ShowFreqsContext *s, int f)
+{
+    switch (s->fscale) {
+    case FS_LINEAR:
+        return (s->w/(float)s->nb_freq)*f;
+    case FS_LOG:
+        return s->w-pow(s->w, (s->nb_freq-f-1)/(s->nb_freq-1.));
+    case FS_RLOG:
+        return pow(s->w, f/(s->nb_freq-1.));
+    }
+
+    return 0;
+}
+
+static float get_bsize(ShowFreqsContext *s, int f)
+{
+    switch (s->fscale) {
+    case FS_LINEAR:
+        return s->w/(float)s->nb_freq;
+    case FS_LOG:
+        return pow(s->w, (s->nb_freq-f-1)/(s->nb_freq-1.))-
+               pow(s->w, (s->nb_freq-f-2)/(s->nb_freq-1.));
+    case FS_RLOG:
+        return pow(s->w, (f+1)/(s->nb_freq-1.))-
+               pow(s->w,  f   /(s->nb_freq-1.));
+    }
+
+    return 1.;
+}
+
+static inline void plot_freq(ShowFreqsContext *s, int ch,
+                             double a, int f, uint8_t fg[4], int *prev_y,
+                             AVFrame *out, AVFilterLink *outlink)
+{
+    const int w = s->w;
+    const float avg = s->avg_data[ch][f];
+    const float bsize = get_bsize(s, f);
+    const int sx = get_sx(s, f);
+    int x, y, i;
+
+    switch(s->ascale) {
+    case AS_SQRT:
+        a = 1.0 - sqrt(a);
+        break;
+    case AS_CBRT:
+        a = 1.0 - cbrt(a);
+        break;
+    case AS_LOG:
+        a = log(av_clipd(a, 1e-6, 1)) / log(1e-6);
+        break;
+    case AS_LINEAR:
+        a = 1.0 - a;
+        break;
+    }
+    y = a * outlink->h - 1;
+    if (y < 0)
+        return;
+
+    switch (s->avg) {
+    case 0:
+        y = s->avg_data[ch][f] = !outlink->frame_count ? y : FFMIN(avg, y);
+        break;
+    case 1:
+        break;
+    default:
+        s->avg_data[ch][f] = avg + y * (y - avg) / (FFMIN(outlink->frame_count + 1, s->avg) * y);
+        y = s->avg_data[ch][f];
+        break;
+    }
+
+    switch(s->mode) {
+    case LINE:
+        if (*prev_y == -1) {
+            *prev_y = y;
+        }
+        if (y <= *prev_y) {
+            for (x = sx + 1; x < sx + bsize && x < w; x++)
+                draw_dot(out, x, y, fg);
+            for (i = y; i <= *prev_y; i++)
+                draw_dot(out, sx, i, fg);
+        } else {
+            for (i = *prev_y; i <= y; i++)
+                draw_dot(out, sx, i, fg);
+            for (x = sx + 1; x < sx + bsize && x < w; x++)
+                draw_dot(out, x, i - 1, fg);
+        }
+        *prev_y = y;
+        break;
+    case BAR:
+        for (x = sx; x < sx + bsize && x < w; x++)
+            for (i = y; i < outlink->h; i++)
+                draw_dot(out, x, i, fg);
+        break;
+    case DOT:
+        for (x = sx; x < sx + bsize && x < w; x++)
+            draw_dot(out, x, y, fg);
+        break;
+    }
+}
+
+static int plot_freqs(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    ShowFreqsContext *s = ctx->priv;
+    const int win_size = s->win_size;
+    char *colors, *color, *saveptr = NULL;
+    AVFrame *out;
+    int ch, n;
+
+    out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
+    if (!out)
+        return AVERROR(ENOMEM);
+
+    for (n = 0; n < outlink->h; n++)
+        memset(out->data[0] + out->linesize[0] * n, 0, outlink->w * 4);
+
+    /* fill FFT input with the number of samples available */
+    for (ch = 0; ch < s->nb_channels; ch++) {
+        const float *p = (float *)in->extended_data[ch];
+
+        for (n = 0; n < in->nb_samples; n++) {
+            s->fft_data[ch][n].re = p[n] * s->window_func_lut[n];
+            s->fft_data[ch][n].im = 0;
+        }
+        for (; n < win_size; n++) {
+            s->fft_data[ch][n].re = 0;
+            s->fft_data[ch][n].im = 0;
+        }
+    }
+
+    /* run FFT on each samples set */
+    for (ch = 0; ch < s->nb_channels; ch++) {
+        av_fft_permute(s->fft, s->fft_data[ch]);
+        av_fft_calc(s->fft, s->fft_data[ch]);
+    }
+
+#define RE(x, ch) s->fft_data[ch][x].re
+#define IM(x, ch) s->fft_data[ch][x].im
+#define M(a, b) (sqrt((a) * (a) + (b) * (b)))
+
+    colors = av_strdup(s->colors);
+    if (!colors)
+        return AVERROR(ENOMEM);
+
+    for (ch = 0; ch < s->nb_channels; ch++) {
+        uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };
+        int prev_y = -1, f;
+        double a;
+
+        color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr);
+        if (color)
+            av_parse_color(fg, color, -1, ctx);
+
+        a = av_clipd(M(RE(0, ch), 0) / s->scale, 0, 1);
+        plot_freq(s, ch, a, 0, fg, &prev_y, out, outlink);
+
+        for (f = 1; f < s->nb_freq; f++) {
+            a = av_clipd(M(RE(f, ch), IM(f, ch)) / s->scale, 0, 1);
+
+            plot_freq(s, ch, a, f, fg, &prev_y, out, outlink);
+        }
+    }
+
+    av_free(colors);
+    out->pts = in->pts;
+    return ff_filter_frame(outlink, out);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ShowFreqsContext *s = ctx->priv;
+    AVFrame *fin = NULL;
+    int ret;
+
+    av_audio_fifo_write(s->fifo, (void **)in->extended_data, in->nb_samples);
+    while (av_audio_fifo_size(s->fifo) >= s->win_size) {
+        fin = ff_get_audio_buffer(inlink, s->win_size);
+        if (!fin) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        fin->pts = s->pts;
+        s->pts += s->skip_samples;
+        ret = av_audio_fifo_peek(s->fifo, (void **)fin->extended_data, s->win_size);
+        if (ret < 0)
+            goto fail;
+
+        ret = plot_freqs(inlink, fin);
+        av_audio_fifo_drain(s->fifo, s->skip_samples);
+        if (ret < 0)
+            goto fail;
+    }
+
+fail:
+    av_frame_free(&fin);
+    av_frame_free(&in);
+    return ret;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ShowFreqsContext *s = ctx->priv;
+    int i;
+
+    av_fft_end(s->fft);
+    for (i = 0; i < s->nb_channels; i++) {
+        av_freep(&s->fft_data[i]);
+        av_freep(&s->avg_data[i]);
+    }
+    av_freep(&s->fft_data);
+    av_freep(&s->avg_data);
+    av_freep(&s->window_func_lut);
+    av_audio_fifo_free(s->fifo);
+}
+
+static const AVFilterPad showfreqs_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad showfreqs_outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_VIDEO,
+        .config_props  = config_output,
+    },
+    { NULL }
+};
+
+AVFilter ff_avf_showfreqs = {
+    .name          = "showfreqs",
+    .description   = NULL_IF_CONFIG_SMALL("Convert input audio to a frequencies video output."),
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .priv_size     = sizeof(ShowFreqsContext),
+    .inputs        = showfreqs_inputs,
+    .outputs       = showfreqs_outputs,
+    .priv_class    = &showfreqs_class,
+};
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 6fc41451ed..d0c01b1b57 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  5
-#define LIBAVFILTER_VERSION_MINOR  34
+#define LIBAVFILTER_VERSION_MINOR  35
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
index 574907aa32..d5298cce4d 100644
--- a/libavutil/audio_fifo.c
+++ b/libavutil/audio_fifo.c
@@ -136,6 +136,25 @@ int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
     return nb_samples;
 }
 
+int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
+{
+    int i, ret, size;
+
+    if (nb_samples < 0)
+        return AVERROR(EINVAL);
+    nb_samples = FFMIN(nb_samples, af->nb_samples);
+    if (!nb_samples)
+        return 0;
+
+    size = nb_samples * af->sample_size;
+    for (i = 0; i < af->nb_buffers; i++) {
+        if ((ret = av_fifo_generic_peek(af->buf[i], data[i], size, NULL)) < 0)
+            return AVERROR_BUG;
+    }
+
+    return nb_samples;
+}
+
 int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
 {
     int i, ret, size;
diff --git a/libavutil/audio_fifo.h b/libavutil/audio_fifo.h
index d21e6a1318..d93be2aeb0 100644
--- a/libavutil/audio_fifo.h
+++ b/libavutil/audio_fifo.h
@@ -94,6 +94,22 @@ int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples);
 int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
 
 /**
+ * Peek data from an AVAudioFifo.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af          AVAudioFifo to read from
+ * @param data        audio data plane pointers
+ * @param nb_samples  number of samples to peek
+ * @return            number of samples actually peek, or negative AVERROR code
+ *                    on failure. The number of samples actually peek will not
+ *                    be greater than nb_samples, and will only be less than
+ *                    nb_samples if av_audio_fifo_size is less than nb_samples.
+ */
+int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples);
+
+/**
  * Read data from an AVAudioFifo.
  *
  * @see enum AVSampleFormat
diff --git a/libavutil/fifo.c b/libavutil/fifo.c
index f2fe93de75..07fb4ec811 100644
--- a/libavutil/fifo.c
+++ b/libavutil/fifo.c
@@ -148,6 +148,32 @@ int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size,
     return total - size;
 }
 
+int av_fifo_generic_peek(AVFifoBuffer *f, void *dest, int buf_size,
+                         void (*func)(void *, void *, int))
+{
+// Read memory barrier needed for SMP here in theory
+    uint8_t *rptr = f->rptr;
+    uint32_t rndx = f->rndx;
+
+    do {
+        int len = FFMIN(f->end - f->rptr, buf_size);
+        if (func)
+            func(dest, f->rptr, len);
+        else {
+            memcpy(dest, f->rptr, len);
+            dest = (uint8_t *)dest + len;
+        }
+// memory barrier needed for SMP here in theory
+        av_fifo_drain(f, len);
+        buf_size -= len;
+    } while (buf_size > 0);
+
+    f->rptr = rptr;
+    f->rndx = rndx;
+
+    return 0;
+}
+
 int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size,
                          void (*func)(void *, void *, int))
 {
diff --git a/libavutil/fifo.h b/libavutil/fifo.h
index f3bdcbceb4..0e4070b99e 100644
--- a/libavutil/fifo.h
+++ b/libavutil/fifo.h
@@ -85,6 +85,16 @@ int av_fifo_space(const AVFifoBuffer *f);
 
 /**
  * Feed data from an AVFifoBuffer to a user-supplied callback.
+ * Similar as av_fifo_gereric_read but without discarding data.
+ * @param f AVFifoBuffer to read from
+ * @param buf_size number of bytes to read
+ * @param func generic read function
+ * @param dest data destination
+ */
+int av_fifo_generic_peek(AVFifoBuffer *f, void *dest, int buf_size, void (*func)(void*, void*, int));
+
+/**
+ * Feed data from an AVFifoBuffer to a user-supplied callback.
  * @param f AVFifoBuffer to read from
  * @param buf_size number of bytes to read
  * @param func generic read function