/* * 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; int bmode_in; int bmode_out; 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, 8, 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" }, { "ms>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, A, "mode" }, { "ms>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=8}, 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 }, { "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" }, { "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" }, { "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" }, { "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" }, { "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" }, { NULL } }; AVFILTER_DEFINE_CLASS(stereotools); 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; StereoToolsContext *s = ctx->priv; s->length = 2 * inlink->sample_rate * 0.05; if (s->length <= 1 || s->length & 1) { av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n"); return AVERROR(EINVAL); } 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 = s->mute_l; const int mute_r = s->mute_r; const int phase_l = s->phase_l; const int phase_r = s->phase_r; double *buffer = s->buffer; AVFrame *out; double *dst; int nbuf = inlink->sample_rate * (fabs(delay) / 1000.); int n; nbuf -= nbuf % 2; if (av_frame_is_writable(in)) { out = in; } else { out = ff_get_audio_buffer(outlink, 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, gl, gr, gd; L *= level_in; R *= level_in; gl = 1. - FFMAX(0., balance_in); gr = 1. + FFMIN(0., balance_in); switch (s->bmode_in) { case 1: gd = gl - gr; gl = 1. + gd; gr = 1. - gd; break; case 2: if (balance_in < 0.) { gr = FFMAX(0.5, gr); gl = 1. / gr; } else if (balance_in > 0.) { gl = FFMAX(0.5, gl); gr = 1. / gl; } break; } L *= gl; R *= gr; 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; case 7: l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal); L = l; R = l; break; case 8: r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal); L = r; 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; gl = 1. - FFMAX(0., balance_out); gr = 1. + FFMIN(0., balance_out); switch (s->bmode_out) { case 1: gd = gl - gr; gl = 1. + gd; gr = 1. - gd; break; case 2: if (balance_out < 0.) { gr = FFMAX(0.5, gr); gl = 1. / gr; } else if (balance_out > 0.) { gl = FFMAX(0.5, gl); gr = 1. / gl; } break; } L *= gl; R *= gr; 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, };