avfilter: add stereo tools filter

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

4 files changed, 305 insertions, 1 deletions

diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 03e18d2da9..05effd6f3f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -80,6 +80,7 @@ OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
 OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER)      += af_sidechaincompress.o
 OBJS-$(CONFIG_SILENCEDETECT_FILTER)          += af_silencedetect.o
 OBJS-$(CONFIG_SILENCEREMOVE_FILTER)          += af_silenceremove.o
+OBJS-$(CONFIG_STEREOTOOLS_FILTER)            += af_stereotools.o
 OBJS-$(CONFIG_STEREOWIDEN_FILTER)            += af_stereowiden.o
 OBJS-$(CONFIG_TREBLE_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_VOLUME_FILTER)                 += af_volume.o
diff --git a/libavfilter/af_stereotools.c b/libavfilter/af_stereotools.c
new file mode 100644
index 0000000000..f4030b7c4e
--- /dev/null
+++ b/libavfilter/af_stereotools.c
@@ -0,0 +1,302 @@
+/*
+ * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
+ *
+ * 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/channel_layout.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct StereoToolsContext {
+    const AVClass *class;
+
+    int softclip;
+    int mute_l;
+    int mute_r;
+    int phase_l;
+    int phase_r;
+    int mode;
+    double slev;
+    double sbal;
+    double mlev;
+    double mpan;
+    double phase;
+    double base;
+    double delay;
+    double balance_in;
+    double balance_out;
+    double phase_sin_coef;
+    double phase_cos_coef;
+    double sc_level;
+    double inv_atan_shape;
+    double level_in;
+    double level_out;
+
+    double *buffer;
+    int length;
+    int pos;
+} StereoToolsContext;
+
+#define OFFSET(x) offsetof(StereoToolsContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption stereotools_options[] = {
+    { "level_in",    "set level in",     OFFSET(level_in),    AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
+    { "level_out",   "set level out",    OFFSET(level_out),   AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
+    { "balance_in",  "set balance in",   OFFSET(balance_in),  AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
+    { "balance_out", "set balance out",  OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
+    { "softclip",    "enable softclip",  OFFSET(softclip),    AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
+    { "mutel",       "mute L",           OFFSET(mute_l),      AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
+    { "muter",       "mute R",           OFFSET(mute_r),      AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
+    { "phasel",      "phase L",          OFFSET(phase_l),     AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
+    { "phaser",      "phase R",          OFFSET(phase_r),     AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
+    { "mode",        "set stereo mode",  OFFSET(mode),        AV_OPT_TYPE_INT,    {.i64=0},   0,          6, A, "mode" },
+    {     "lr>lr",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=0},   0,          0, A, "mode" },
+    {     "lr>ms",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=1},   0,          0, A, "mode" },
+    {     "ms>lr",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=2},   0,          0, A, "mode" },
+    {     "lr>ll",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=3},   0,          0, A, "mode" },
+    {     "lr>rr",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=4},   0,          0, A, "mode" },
+    {     "lr>l+r",  0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=5},   0,          0, A, "mode" },
+    {     "lr>rl",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=6},   0,          0, A, "mode" },
+    { "slev",        "set side level",   OFFSET(slev),        AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
+    { "sbal",        "set side balance", OFFSET(sbal),        AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
+    { "mlev",        "set middle level", OFFSET(mlev),        AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
+    { "mpan",        "set middle pan",   OFFSET(mpan),        AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
+    { "base",        "set stereo base",  OFFSET(base),        AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
+    { "delay",       "set delay",        OFFSET(delay),       AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20,         20, A },
+    { "sclevel",     "set S/C level",    OFFSET(sc_level),    AV_OPT_TYPE_DOUBLE, {.dbl=1},   1,        100, A },
+    { "phase",       "set stereo phase", OFFSET(phase),       AV_OPT_TYPE_DOUBLE, {.dbl=0},   0,        360, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(stereotools);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layout = NULL;
+
+    ff_add_format(&formats, AV_SAMPLE_FMT_DBL);
+    ff_set_common_formats(ctx, formats);
+    ff_add_channel_layout(&layout, AV_CH_LAYOUT_STEREO);
+    ff_set_common_channel_layouts(ctx, layout);
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    StereoToolsContext *s = ctx->priv;
+
+    s->length = 2 * inlink->sample_rate * 0.05;
+    s->buffer = av_calloc(s->length, sizeof(*s->buffer));
+    if (!s->buffer)
+        return AVERROR(ENOMEM);
+
+    s->inv_atan_shape = 1.0 / atan(s->sc_level);
+    s->phase_cos_coef = cos(s->phase / 180 * M_PI);
+    s->phase_sin_coef = sin(s->phase / 180 * M_PI);
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    StereoToolsContext *s = ctx->priv;
+    const double *src = (const double *)in->data[0];
+    const double sb = s->base < 0 ? s->base * 0.5 : s->base;
+    const double sbal = 1 + s->sbal;
+    const double mpan = 1 + s->mpan;
+    const double slev = s->slev;
+    const double mlev = s->mlev;
+    const double balance_in = s->balance_in;
+    const double balance_out = s->balance_out;
+    const double level_in = s->level_in;
+    const double level_out = s->level_out;
+    const double sc_level = s->sc_level;
+    const double delay = s->delay;
+    const int length = s->length;
+    const int mute_l = floor(s->mute_l + 0.5);
+    const int mute_r = floor(s->mute_r + 0.5);
+    const int phase_l = floor(s->phase_l + 0.5);
+    const int phase_r = floor(s->phase_r + 0.5);
+    double *buffer = s->buffer;
+    AVFrame *out = NULL;
+    double *dst;
+    int nbuf = inlink->sample_rate * (FFABS(delay) / 1000.);
+    int n;
+
+    nbuf -= nbuf % 2;
+    if (av_frame_is_writable(in)) {
+        out = in;
+    } else {
+        AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples);
+        if (!out) {
+            av_frame_free(&in);
+            return AVERROR(ENOMEM);
+        }
+        av_frame_copy_props(out, in);
+    }
+    dst = (double *)out->data[0];
+
+    for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
+        double L = src[0], R = src[1], l, r, m, S;
+
+        L *= level_in;
+        R *= level_in;
+
+        L *= 1. - FFMAX(0., balance_in);
+        R *= 1. + FFMIN(0., balance_in);
+
+        if (s->softclip) {
+            R = s->inv_atan_shape * atan(R * sc_level);
+            L = s->inv_atan_shape * atan(L * sc_level);
+        }
+
+        switch (s->mode) {
+        case 0:
+            m = (L + R) * 0.5;
+            S = (L - R) * 0.5;
+            l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
+            r = m * mlev * FFMIN(1., mpan)      - S * slev * FFMIN(1., sbal);
+            L = l;
+            R = r;
+            break;
+        case 1:
+            l = L * FFMIN(1., 2. - sbal);
+            r = R * FFMIN(1., sbal);
+            L = 0.5 * (l + r) * mlev;
+            R = 0.5 * (l - r) * slev;
+            break;
+        case 2:
+            l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
+            r = L * mlev * FFMIN(1., mpan)      - R * slev * FFMIN(1., sbal);
+            L = l;
+            R = r;
+            break;
+        case 3:
+            R = L;
+            break;
+        case 4:
+            L = R;
+            break;
+        case 5:
+            L = (L + R) / 2;
+            R = L;
+            break;
+        case 6:
+            l = L;
+            L = R;
+            R = l;
+            m = (L + R) * 0.5;
+            S = (L - R) * 0.5;
+            l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
+            r = m * mlev * FFMIN(1., mpan)      - S * slev * FFMIN(1., sbal);
+            L = l;
+            R = r;
+            break;
+        }
+
+        L *= 1. - mute_l;
+        R *= 1. - mute_r;
+
+        L *= (2. * (1. - phase_l)) - 1.;
+        R *= (2. * (1. - phase_r)) - 1.;
+
+        buffer[s->pos  ] = L;
+        buffer[s->pos+1] = R;
+
+        if (delay > 0.) {
+            R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
+        } else if (delay < 0.) {
+            L = buffer[(s->pos - (int)nbuf + length)     % length];
+        }
+
+        l = L + sb * L - sb * R;
+        r = R + sb * R - sb * L;
+
+        L = l;
+        R = r;
+
+        l = L * s->phase_cos_coef - R * s->phase_sin_coef;
+        r = L * s->phase_sin_coef + R * s->phase_cos_coef;
+
+        L = l;
+        R = r;
+
+        s->pos = (s->pos + 2) % s->length;
+
+        L *= 1. - FFMAX(0., balance_out);
+        R *= 1. + FFMIN(0., balance_out);
+
+        L *= level_out;
+        R *= level_out;
+
+        dst[0] = L;
+        dst[1] = R;
+    }
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    StereoToolsContext *s = ctx->priv;
+
+    av_freep(&s->buffer);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_stereotools = {
+    .name           = "stereotools",
+    .description    = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(StereoToolsContext),
+    .priv_class     = &stereotools_class,
+    .uninit         = uninit,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 89390aaaa0..cab4564bd5 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -102,6 +102,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(SIDECHAINCOMPRESS, sidechaincompress, af);
     REGISTER_FILTER(SILENCEDETECT,  silencedetect,  af);
     REGISTER_FILTER(SILENCEREMOVE,  silenceremove,  af);
+    REGISTER_FILTER(STEREOTOOLS,    stereotools,    af);
     REGISTER_FILTER(STEREOWIDEN,    stereowiden,    af);
     REGISTER_FILTER(TREBLE,         treble,         af);
     REGISTER_FILTER(VOLUME,         volume,         af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index b3ad51a52a..e918184f6a 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   4
+#define LIBAVFILTER_VERSION_MINOR   5
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \