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Diffstat (limited to 'libavfilter/af_sidechaincompress.c')
| -rw-r--r-- | libavfilter/af_sidechaincompress.c | 450 |
1 files changed, 450 insertions, 0 deletions
diff --git a/libavfilter/af_sidechaincompress.c b/libavfilter/af_sidechaincompress.c new file mode 100644 index 0000000000..3f540e2dff --- /dev/null +++ b/libavfilter/af_sidechaincompress.c @@ -0,0 +1,450 @@ +/* + * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others + * 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 + */ + +/** + * @file + * Audio (Sidechain) Compressor filter + */ + +#include "libavutil/audio_fifo.h" +#include "libavutil/avassert.h" +#include "libavutil/channel_layout.h" +#include "libavutil/common.h" +#include "libavutil/opt.h" + +#include "audio.h" +#include "avfilter.h" +#include "formats.h" +#include "hermite.h" +#include "internal.h" + +typedef struct SidechainCompressContext { + const AVClass *class; + + double level_in; + double level_sc; + double attack, attack_coeff; + double release, release_coeff; + double lin_slope; + double ratio; + double threshold; + double makeup; + double mix; + double thres; + double knee; + double knee_start; + double knee_stop; + double lin_knee_start; + double adj_knee_start; + double compressed_knee_stop; + int link; + int detection; + + AVAudioFifo *fifo[2]; + int64_t pts; +} SidechainCompressContext; + +#define OFFSET(x) offsetof(SidechainCompressContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM +#define F AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption options[] = { + { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F }, + { "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563, 1, A|F }, + { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 20, A|F }, + { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, A|F }, + { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A|F }, + { "makeup", "set make up gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 64, A|F }, + { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.82843}, 1, 8, A|F }, + { "link", "set link type", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A|F, "link" }, + { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "link" }, + { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "link" }, + { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A|F, "detection" }, + { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "detection" }, + { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "detection" }, + { "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F }, + { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A|F }, + { NULL } +}; + +#define sidechaincompress_options options +AVFILTER_DEFINE_CLASS(sidechaincompress); + +// A fake infinity value (because real infinity may break some hosts) +#define FAKE_INFINITY (65536.0 * 65536.0) + +// Check for infinity (with appropriate-ish tolerance) +#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0) + +static double output_gain(double lin_slope, double ratio, double thres, + double knee, double knee_start, double knee_stop, + double compressed_knee_stop, int detection) +{ + double slope = log(lin_slope); + double gain = 0.0; + double delta = 0.0; + + if (detection) + slope *= 0.5; + + if (IS_FAKE_INFINITY(ratio)) { + gain = thres; + delta = 0.0; + } else { + gain = (slope - thres) / ratio + thres; + delta = 1.0 / ratio; + } + + if (knee > 1.0 && slope < knee_stop) + gain = hermite_interpolation(slope, knee_start, knee_stop, + knee_start, compressed_knee_stop, + 1.0, delta); + + return exp(gain - slope); +} + +static int compressor_config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + SidechainCompressContext *s = ctx->priv; + + s->thres = log(s->threshold); + s->lin_knee_start = s->threshold / sqrt(s->knee); + s->adj_knee_start = s->lin_knee_start * s->lin_knee_start; + s->knee_start = log(s->lin_knee_start); + s->knee_stop = log(s->threshold * sqrt(s->knee)); + s->compressed_knee_stop = (s->knee_stop - s->thres) / s->ratio + s->thres; + + s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.)); + s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.)); + + return 0; +} + +static void compressor(SidechainCompressContext *s, + const double *src, double *dst, const double *scsrc, int nb_samples, + double level_in, double level_sc, + AVFilterLink *inlink, AVFilterLink *sclink) +{ + const double makeup = s->makeup; + const double mix = s->mix; + int i, c; + + for (i = 0; i < nb_samples; i++) { + double abs_sample, gain = 1.0; + + abs_sample = fabs(scsrc[0] * level_sc); + + if (s->link == 1) { + for (c = 1; c < sclink->channels; c++) + abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample); + } else { + for (c = 1; c < sclink->channels; c++) + abs_sample += fabs(scsrc[c] * level_sc); + + abs_sample /= sclink->channels; + } + + if (s->detection) + abs_sample *= abs_sample; + + s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? s->attack_coeff : s->release_coeff); + + if (s->lin_slope > 0.0 && s->lin_slope > (s->detection ? s->adj_knee_start : s->lin_knee_start)) + gain = output_gain(s->lin_slope, s->ratio, s->thres, s->knee, + s->knee_start, s->knee_stop, + s->compressed_knee_stop, s->detection); + + for (c = 0; c < inlink->channels; c++) + dst[c] = src[c] * level_in * (gain * makeup * mix + (1. - mix)); + + src += inlink->channels; + dst += inlink->channels; + scsrc += sclink->channels; + } +} + +#if CONFIG_SIDECHAINCOMPRESS_FILTER +static int filter_frame(AVFilterLink *link, AVFrame *frame) +{ + AVFilterContext *ctx = link->dst; + SidechainCompressContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + AVFrame *out = NULL, *in[2] = { NULL }; + double *dst; + int nb_samples; + int i; + + for (i = 0; i < 2; i++) + if (link == ctx->inputs[i]) + break; + av_assert0(i < 2); + av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data, + frame->nb_samples); + av_frame_free(&frame); + + nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1])); + if (!nb_samples) + return 0; + + out = ff_get_audio_buffer(outlink, nb_samples); + if (!out) + return AVERROR(ENOMEM); + for (i = 0; i < 2; i++) { + in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples); + if (!in[i]) { + av_frame_free(&in[0]); + av_frame_free(&in[1]); + av_frame_free(&out); + return AVERROR(ENOMEM); + } + av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples); + } + + dst = (double *)out->data[0]; + out->pts = s->pts; + s->pts += nb_samples; + + compressor(s, (double *)in[0]->data[0], dst, + (double *)in[1]->data[0], nb_samples, + s->level_in, s->level_sc, + ctx->inputs[0], ctx->inputs[1]); + + av_frame_free(&in[0]); + av_frame_free(&in[1]); + + return ff_filter_frame(outlink, out); +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + SidechainCompressContext *s = ctx->priv; + int i; + + /* get a frame on each input */ + for (i = 0; i < 2; i++) { + AVFilterLink *inlink = ctx->inputs[i]; + if (!av_audio_fifo_size(s->fifo[i])) + return ff_request_frame(inlink); + } + + return 0; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBL, + AV_SAMPLE_FMT_NONE + }; + int ret, i; + + if (!ctx->inputs[0]->in_channel_layouts || + !ctx->inputs[0]->in_channel_layouts->nb_channel_layouts) { + av_log(ctx, AV_LOG_WARNING, + "No channel layout for input 1\n"); + return AVERROR(EAGAIN); + } + + if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 || + (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0) + return ret; + + for (i = 0; i < 2; i++) { + layouts = ff_all_channel_counts(); + if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0) + return ret; + } + + formats = ff_make_format_list(sample_fmts); + if ((ret = ff_set_common_formats(ctx, formats)) < 0) + return ret; + + formats = ff_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + SidechainCompressContext *s = ctx->priv; + + if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) { + av_log(ctx, AV_LOG_ERROR, + "Inputs must have the same sample rate " + "%d for in0 vs %d for in1\n", + ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate); + return AVERROR(EINVAL); + } + + outlink->sample_rate = ctx->inputs[0]->sample_rate; + outlink->time_base = ctx->inputs[0]->time_base; + outlink->channel_layout = ctx->inputs[0]->channel_layout; + outlink->channels = ctx->inputs[0]->channels; + + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024); + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024); + if (!s->fifo[0] || !s->fifo[1]) + return AVERROR(ENOMEM); + + compressor_config_output(outlink); + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + SidechainCompressContext *s = ctx->priv; + + av_audio_fifo_free(s->fifo[0]); + av_audio_fifo_free(s->fifo[1]); +} + +static const AVFilterPad sidechaincompress_inputs[] = { + { + .name = "main", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + },{ + .name = "sidechain", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad sidechaincompress_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + .request_frame = request_frame, + }, + { NULL } +}; + +AVFilter ff_af_sidechaincompress = { + .name = "sidechaincompress", + .description = NULL_IF_CONFIG_SMALL("Sidechain compressor."), + .priv_size = sizeof(SidechainCompressContext), + .priv_class = &sidechaincompress_class, + .query_formats = query_formats, + .uninit = uninit, + .inputs = sidechaincompress_inputs, + .outputs = sidechaincompress_outputs, +}; +#endif /* CONFIG_SIDECHAINCOMPRESS_FILTER */ + +#if CONFIG_ACOMPRESSOR_FILTER +static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + const double *src = (const double *)in->data[0]; + AVFilterContext *ctx = inlink->dst; + SidechainCompressContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + AVFrame *out; + double *dst; + + 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]; + + compressor(s, src, dst, src, in->nb_samples, + s->level_in, s->level_in, + inlink, inlink); + + if (out != in) + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static int acompressor_query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBL, + AV_SAMPLE_FMT_NONE + }; + int ret; + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +#define acompressor_options options +AVFILTER_DEFINE_CLASS(acompressor); + +static const AVFilterPad acompressor_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = acompressor_filter_frame, + }, + { NULL } +}; + +static const AVFilterPad acompressor_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = compressor_config_output, + }, + { NULL } +}; + +AVFilter ff_af_acompressor = { + .name = "acompressor", + .description = NULL_IF_CONFIG_SMALL("Audio compressor."), + .priv_size = sizeof(SidechainCompressContext), + .priv_class = &acompressor_class, + .query_formats = acompressor_query_formats, + .inputs = acompressor_inputs, + .outputs = acompressor_outputs, +}; +#endif /* CONFIG_ACOMPRESSOR_FILTER */ |