[FFmpeg-devel] avfilter: add Haas stereo enhancer

Submitted by Paul B Mahol on Sept. 6, 2017, 4:50 p.m.

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Message ID 20170906165015.15673-1-onemda@gmail.com
State New
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Paul B Mahol Sept. 6, 2017, 4:50 p.m.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi         |  64 ++++++++++++++
 libavfilter/Makefile     |   1 +
 libavfilter/af_haas.c    | 226 +++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 292 insertions(+)
 create mode 100644 libavfilter/af_haas.c

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diff --git a/doc/filters.texi b/doc/filters.texi
index 7790367..c3c54fd 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2770,6 +2770,70 @@  Set delay-line interpolation, @var{linear} or @var{quadratic}.
 Default is @var{linear}.
 @end table
 
+@section haas
+Apply Haas effect to audio.
+
+Note that this makes most sense to apply on mono signals.
+With this filter applied to mono signals it give some directionality and
+streches its stereo image.
+
+The filter accepts the following options:
+
+@table @option
+@item level_in
+Set input level. By default is @var{1}, or 0dB
+
+@item level_out
+Set output level. By default is @var{1}, or 0dB.
+
+@item side_gain
+Set gain applied to side part of signal. By default is @var{1}.
+
+@item middle_source
+Set kind of middle source. Can be one of the following:
+
+@table @samp
+@item left
+Pick left channel.
+
+@item right
+Pick right channel.
+
+@item mid
+Pick middle part signal of stereo image.
+
+@item side
+Pick side part signal of stereo image.
+@end table
+
+@item middle_phase
+Change middle phase. By default is disabled.
+
+@item left_delay
+Set left channel delay. By default is @var{2.05} milliseconds.
+
+@item left_balance
+Set left channel balance. By default is @var{-1}.
+
+@item left_gain
+Set left channel gain. By default is @var{1}.
+
+@item left_phase
+Change left phase. By default is disabled.
+
+@item right_delay
+Set right channel delay. By defaults is @var{2.12} milliseconds.
+
+@item right_balance
+Set right channel balance. By default is @var{1}.
+
+@item right_gain
+Set right channel gain. By default is @var{1}.
+
+@item right_phase
+Change right phase. By default is enabled.
+@end table
+
 @section hdcd
 
 Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 1e460ab..4268633 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -91,6 +91,7 @@  OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o
 OBJS-$(CONFIG_EXTRASTEREO_FILTER)            += af_extrastereo.o
 OBJS-$(CONFIG_FIREQUALIZER_FILTER)           += af_firequalizer.o
 OBJS-$(CONFIG_FLANGER_FILTER)                += af_flanger.o generate_wave_table.o
+OBJS-$(CONFIG_HAAS_FILTER)                   += af_haas.o
 OBJS-$(CONFIG_HDCD_FILTER)                   += af_hdcd.o
 OBJS-$(CONFIG_HEADPHONE_FILTER)              += af_headphone.o
 OBJS-$(CONFIG_HIGHPASS_FILTER)               += af_biquads.o
diff --git a/libavfilter/af_haas.c b/libavfilter/af_haas.c
new file mode 100644
index 0000000..a9761bd
--- /dev/null
+++ b/libavfilter/af_haas.c
@@ -0,0 +1,226 @@ 
+/*
+ * Copyright (c) 2001-2010 Vladimir Sadovnikov
+ *
+ * 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 HaasContext {
+    const AVClass *class;
+
+    int par_m_source;
+    int par_delay0;
+    int par_delay1;
+    int par_phase0;
+    int par_phase1;
+    int par_middle_phase;
+    double par_side_gain;
+    double par_gain0;
+    double par_gain1;
+    double par_balance0;
+    double par_balance1;
+    double level_in;
+    double level_out;
+
+    double *buffer;
+    int buffer_size;
+    uint32_t write_ptr;
+    uint32_t delay[2];
+    double balance_l[2];
+    double balance_r[2];
+    double phase0;
+    double phase1;
+} HaasContext;
+
+#define OFFSET(x) offsetof(HaasContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption haas_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 },
+    { "side_gain",     "set side gain",     OFFSET(par_side_gain),    AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
+    { "middle_source", "set middle source", OFFSET(par_m_source),     AV_OPT_TYPE_INT,     {.i64=2},        0,   3, A, "source" },
+    {   "left",        0,                   0,                        AV_OPT_TYPE_CONST,   {.i64=0},        0,   0, A, "source" },
+    {   "right",       0,                   0,                        AV_OPT_TYPE_CONST,   {.i64=1},        0,   0, A, "source" },
+    {   "mid",         "L+R",               0,                        AV_OPT_TYPE_CONST,   {.i64=2},        0,   0, A, "source" },
+    {   "side",        "L-R",               0,                        AV_OPT_TYPE_CONST,   {.i64=3},        0,   0, A, "source" },
+    { "middle_phase",  "set middle phase",  OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL,    {.i64=0},        0,   1, A },
+    { "left_delay",    "set left delay",    OFFSET(par_delay0),       AV_OPT_TYPE_DOUBLE,  {.dbl=2.05},     0,  40, A },
+    { "left_balance",  "set left balance",  OFFSET(par_balance0),     AV_OPT_TYPE_DOUBLE,  {.dbl=-1.0},    -1,   1, A },
+    { "left_gain",     "set left gain",     OFFSET(par_gain0),        AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
+    { "left_phase",    "set left phase",    OFFSET(par_phase0),       AV_OPT_TYPE_BOOL,    {.i64=0},        0,   1, A },
+    { "right_delay",   "set right delay",   OFFSET(par_delay1),       AV_OPT_TYPE_DOUBLE,  {.dbl=2.12},     0,  40, A },
+    { "right_balance", "set right balance", OFFSET(par_balance1),     AV_OPT_TYPE_DOUBLE,  {.dbl=1},       -1,   1, A },
+    { "right_gain",    "set right gain",    OFFSET(par_gain1),        AV_OPT_TYPE_DOUBLE,  {.dbl=1}, 0.015625,  64, A },
+    { "right_phase",   "set right phase",   OFFSET(par_phase1),       AV_OPT_TYPE_BOOL,    {.i64=1},        0,   1, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(haas);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layout = NULL;
+    int ret;
+
+    if ((ret = ff_add_format                 (&formats, AV_SAMPLE_FMT_DBL  )) < 0 ||
+        (ret = ff_set_common_formats         (ctx     , formats            )) < 0 ||
+        (ret = ff_add_channel_layout         (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
+        (ret = ff_set_common_channel_layouts (ctx     , layout             )) < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    HaasContext *s = ctx->priv;
+    size_t min_buf_size = (size_t)(inlink->sample_rate * 10 * 0.001);
+    size_t new_buf_size = 1;
+
+    while (new_buf_size < min_buf_size)
+        new_buf_size <<= 1;
+
+    av_freep(&s->buffer);
+    s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
+    if (!s->buffer)
+        return AVERROR(ENOMEM);
+
+    s->buffer_size = new_buf_size;
+    s->write_ptr = 0;
+
+    s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
+    s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);
+
+    s->phase0 = s->par_phase0 ? 1.0 : -1.0;
+    s->phase1 = s->par_phase1 ? 1.0 : -1.0;
+
+    s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
+    s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
+    s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
+    s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    HaasContext *s = ctx->priv;
+    const double *src = (const double *)in->data[0];
+    const double level_in = s->level_in;
+    const double level_out = s->level_out;
+    const uint32_t mask = s->buffer_size - 1;
+    double *buffer = s->buffer;
+    AVFrame *out;
+    double *dst;
+    int n;
+
+    if (av_frame_is_writable(in)) {
+        out = in;
+    } else {
+        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 mid, side[2], side_l, side_r;
+        uint32_t s0_ptr, s1_ptr;
+
+        switch (s->par_m_source) {
+        case 0: mid = src[0]; break;
+        case 1: mid = src[1]; break;
+        case 2: mid = (src[0] + src[1]) * 0.5; break;
+        case 3: mid = (src[0] - src[1]) * 0.5; break;
+        }
+
+        buffer[s->write_ptr] = mid;
+
+        mid *= level_in;
+
+        s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
+        s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;
+
+        if (s->par_middle_phase)
+            mid = -mid;
+
+        side[0] = buffer[s0_ptr] * s->par_side_gain;
+        side[1] = buffer[s1_ptr] * s->par_side_gain;
+        side_l  = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
+        side_r  = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];
+
+        dst[0] = (mid + side_l) * level_out;
+        dst[1] = (mid + side_r) * level_out;
+
+        s->write_ptr = (s->write_ptr + 1) & mask;
+    }
+
+    if (out != in)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    HaasContext *s = ctx->priv;
+
+    av_freep(&s->buffer);
+    s->buffer_size = 0;
+}
+
+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_haas = {
+    .name           = "haas",
+    .description    = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(HaasContext),
+    .priv_class     = &haas_class,
+    .uninit         = uninit,
+    .inputs         = inputs,
+    .outputs        = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 9a2cfea..9bbc6d6 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -104,6 +104,7 @@  static void register_all(void)
     REGISTER_FILTER(EXTRASTEREO,    extrastereo,    af);
     REGISTER_FILTER(FIREQUALIZER,   firequalizer,   af);
     REGISTER_FILTER(FLANGER,        flanger,        af);
+    REGISTER_FILTER(HAAS,           haas,           af);
     REGISTER_FILTER(HDCD,           hdcd,           af);
     REGISTER_FILTER(HEADPHONE,      headphone,      af);
     REGISTER_FILTER(HIGHPASS,       highpass,       af);