@@ -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
@@ -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
new file mode 100644
@@ -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,
+};
@@ -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);
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