mirror of https://github.com/FFmpeg/FFmpeg.git
parent
3cd8aaa2b2
commit
6b68e2a43b
6 changed files with 597 additions and 2 deletions
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/*
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* Copyright (c) 1999 Chris Bagwell |
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* Copyright (c) 1999 Nick Bailey |
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* Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net> |
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* Copyright (c) 2013 Paul B Mahol |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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* |
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*/ |
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#include "libavutil/avstring.h" |
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#include "libavutil/opt.h" |
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#include "libavutil/samplefmt.h" |
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#include "avfilter.h" |
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#include "audio.h" |
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#include "internal.h" |
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typedef struct ChanParam { |
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double attack; |
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double decay; |
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double volume; |
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} ChanParam; |
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typedef struct CompandSegment { |
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double x, y; |
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double a, b; |
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} CompandSegment; |
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typedef struct CompandContext { |
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const AVClass *class; |
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char *attacks, *decays, *points; |
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CompandSegment *segments; |
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ChanParam *channels; |
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double in_min_lin; |
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double out_min_lin; |
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double curve_dB; |
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double gain_dB; |
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double initial_volume; |
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double delay; |
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uint8_t **delayptrs; |
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int delay_samples; |
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int delay_count; |
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int delay_index; |
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int64_t pts; |
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int (*compand)(AVFilterContext *ctx, AVFrame *frame); |
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} CompandContext; |
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#define OFFSET(x) offsetof(CompandContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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static const AVOption compand_options[] = { |
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{ "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, {.dbl=0.01}, 0.01, 900, A }, |
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{ "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, A }, |
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{ "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 0, A }, |
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{ "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 20, A }, |
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{ NULL }, |
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}; |
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AVFILTER_DEFINE_CLASS(compand); |
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static av_cold int init(AVFilterContext *ctx) |
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{ |
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CompandContext *s = ctx->priv; |
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if (!s->attacks || !s->decays || !s->points) { |
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av_log(ctx, AV_LOG_ERROR, "Missing attacks and/or decays and/or points.\n"); |
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return AVERROR(EINVAL); |
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} |
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return 0; |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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CompandContext *s = ctx->priv; |
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av_freep(&s->channels); |
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av_freep(&s->segments); |
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if (s->delayptrs) |
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av_freep(&s->delayptrs[0]); |
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av_freep(&s->delayptrs); |
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} |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterChannelLayouts *layouts; |
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AVFilterFormats *formats; |
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static const enum AVSampleFormat sample_fmts[] = { |
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AV_SAMPLE_FMT_DBLP, |
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AV_SAMPLE_FMT_NONE |
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}; |
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layouts = ff_all_channel_layouts(); |
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if (!layouts) |
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return AVERROR(ENOMEM); |
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ff_set_common_channel_layouts(ctx, layouts); |
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formats = ff_make_format_list(sample_fmts); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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ff_set_common_formats(ctx, formats); |
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formats = ff_all_samplerates(); |
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if (!formats) |
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return AVERROR(ENOMEM); |
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ff_set_common_samplerates(ctx, formats); |
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return 0; |
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} |
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static void count_items(char *item_str, int *nb_items) |
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{ |
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char *p; |
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*nb_items = 1; |
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for (p = item_str; *p; p++) { |
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if (*p == ' ') |
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(*nb_items)++; |
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} |
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} |
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static void update_volume(ChanParam *cp, double in) |
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{ |
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double delta = in - cp->volume; |
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if (delta > 0.0) |
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cp->volume += delta * cp->attack; |
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else |
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cp->volume += delta * cp->decay; |
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} |
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static double get_volume(CompandContext *s, double in_lin) |
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{ |
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CompandSegment *cs; |
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double in_log, out_log; |
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int i; |
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if (in_lin < s->in_min_lin) |
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return s->out_min_lin; |
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in_log = log(in_lin); |
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for (i = 1;; i++) |
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if (in_log <= s->segments[i + 1].x) |
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break; |
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cs = &s->segments[i]; |
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in_log -= cs->x; |
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out_log = cs->y + in_log * (cs->a * in_log + cs->b); |
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return exp(out_log); |
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} |
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static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame) |
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{ |
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CompandContext *s = ctx->priv; |
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AVFilterLink *inlink = ctx->inputs[0]; |
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const int channels = inlink->channels; |
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const int nb_samples = frame->nb_samples; |
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AVFrame *out_frame; |
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int chan, i; |
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if (av_frame_is_writable(frame)) { |
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out_frame = frame; |
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} else { |
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out_frame = ff_get_audio_buffer(inlink, nb_samples); |
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if (!out_frame) |
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return AVERROR(ENOMEM); |
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av_frame_copy_props(out_frame, frame); |
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} |
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for (chan = 0; chan < channels; chan++) { |
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const double *src = (double *)frame->data[chan]; |
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double *dst = (double *)out_frame->data[chan]; |
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ChanParam *cp = &s->channels[chan]; |
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for (i = 0; i < nb_samples; i++) { |
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update_volume(cp, fabs(src[i])); |
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dst[i] = av_clipd(src[i] * get_volume(s, cp->volume), -1, 1); |
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} |
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} |
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if (frame != out_frame) |
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av_frame_free(&frame); |
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return ff_filter_frame(ctx->outputs[0], out_frame); |
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} |
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#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) |
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static int compand_delay(AVFilterContext *ctx, AVFrame *frame) |
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{ |
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CompandContext *s = ctx->priv; |
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AVFilterLink *inlink = ctx->inputs[0]; |
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const int channels = inlink->channels; |
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const int nb_samples = frame->nb_samples; |
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int chan, i, dindex, oindex, count; |
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AVFrame *out_frame = NULL; |
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for (chan = 0; chan < channels; chan++) { |
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const double *src = (double *)frame->data[chan]; |
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double *dbuf = (double *)s->delayptrs[chan]; |
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ChanParam *cp = &s->channels[chan]; |
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double *dst; |
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count = s->delay_count; |
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dindex = s->delay_index; |
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for (i = 0, oindex = 0; i < nb_samples; i++) { |
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const double in = src[i]; |
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update_volume(cp, fabs(in)); |
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if (count >= s->delay_samples) { |
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if (!out_frame) { |
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out_frame = ff_get_audio_buffer(inlink, nb_samples - i); |
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if (!out_frame) |
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return AVERROR(ENOMEM); |
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av_frame_copy_props(out_frame, frame); |
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out_frame->pts = s->pts; |
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s->pts += av_rescale_q(nb_samples - i, (AVRational){1, inlink->sample_rate}, inlink->time_base); |
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} |
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dst = (double *)out_frame->data[chan]; |
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dst[oindex++] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1); |
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} else { |
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count++; |
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} |
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dbuf[dindex] = in; |
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dindex = MOD(dindex + 1, s->delay_samples); |
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} |
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} |
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s->delay_count = count; |
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s->delay_index = dindex; |
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av_frame_free(&frame); |
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return out_frame ? ff_filter_frame(ctx->outputs[0], out_frame) : 0; |
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} |
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static int compand_drain(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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CompandContext *s = ctx->priv; |
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const int channels = outlink->channels; |
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int chan, i, dindex; |
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AVFrame *frame = NULL; |
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frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count)); |
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if (!frame) |
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return AVERROR(ENOMEM); |
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frame->pts = s->pts; |
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s->pts += av_rescale_q(frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
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for (chan = 0; chan < channels; chan++) { |
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double *dbuf = (double *)s->delayptrs[chan]; |
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double *dst = (double *)frame->data[chan]; |
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ChanParam *cp = &s->channels[chan]; |
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dindex = s->delay_index; |
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for (i = 0; i < frame->nb_samples; i++) { |
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dst[i] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1); |
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dindex = MOD(dindex + 1, s->delay_samples); |
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} |
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} |
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s->delay_count -= frame->nb_samples; |
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s->delay_index = dindex; |
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return ff_filter_frame(outlink, frame); |
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} |
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static int config_output(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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CompandContext *s = ctx->priv; |
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const int sample_rate = outlink->sample_rate; |
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double radius = s->curve_dB * M_LN10 / 20; |
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int nb_attacks, nb_decays, nb_points; |
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char *p, *saveptr = NULL; |
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int new_nb_items, num; |
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int i; |
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count_items(s->attacks, &nb_attacks); |
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count_items(s->decays, &nb_decays); |
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count_items(s->points, &nb_points); |
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if ((nb_attacks > outlink->channels) || (nb_decays > outlink->channels)) { |
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av_log(ctx, AV_LOG_ERROR, "Number of attacks/decays bigger than number of channels.\n"); |
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return AVERROR(EINVAL); |
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} |
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uninit(ctx); |
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s->channels = av_mallocz_array(outlink->channels, sizeof(*s->channels)); |
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s->segments = av_mallocz_array((nb_points + 4) * 2, sizeof(*s->segments)); |
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if (!s->channels || !s->segments) |
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return AVERROR(ENOMEM); |
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p = s->attacks; |
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for (i = 0, new_nb_items = 0; i < nb_attacks; i++) { |
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char *tstr = av_strtok(p, " ", &saveptr); |
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p = NULL; |
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new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1; |
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if (s->channels[i].attack < 0) |
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return AVERROR(EINVAL); |
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} |
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nb_attacks = new_nb_items; |
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p = s->decays; |
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for (i = 0, new_nb_items = 0; i < nb_decays; i++) { |
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char *tstr = av_strtok(p, " ", &saveptr); |
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p = NULL; |
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new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1; |
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if (s->channels[i].decay < 0) |
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return AVERROR(EINVAL); |
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} |
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nb_decays = new_nb_items; |
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if (nb_attacks != nb_decays) { |
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av_log(ctx, AV_LOG_ERROR, "Number of attacks %d differs from number of decays %d.\n", nb_attacks, nb_decays); |
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return AVERROR(EINVAL); |
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} |
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#define S(x) s->segments[2 * ((x) + 1)] |
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p = s->points; |
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for (i = 0, new_nb_items = 0; i < nb_points; i++) { |
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char *tstr = av_strtok(p, " ", &saveptr); |
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p = NULL; |
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if (sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) { |
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av_log(ctx, AV_LOG_ERROR, "Invalid and/or missing input/output value.\n"); |
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return AVERROR(EINVAL); |
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} |
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if (i && S(i - 1).x > S(i).x) { |
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av_log(ctx, AV_LOG_ERROR, "Transfer function input values must be increasing.\n"); |
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return AVERROR(EINVAL); |
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} |
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S(i).y -= S(i).x; |
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av_log(ctx, AV_LOG_DEBUG, "%d: x=%lf y=%lf\n", i, S(i).x, S(i).y); |
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new_nb_items++; |
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} |
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num = new_nb_items; |
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/* Add 0,0 if necessary */ |
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if (num == 0 || S(num - 1).x) |
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num++; |
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#undef S |
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#define S(x) s->segments[2 * (x)] |
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/* Add a tail off segment at the start */ |
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S(0).x = S(1).x - 2 * s->curve_dB; |
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S(0).y = S(1).y; |
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num++; |
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/* Join adjacent colinear segments */ |
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for (i = 2; i < num; i++) { |
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double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x); |
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double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x); |
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int j; |
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if (fabs(g1 - g2)) |
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continue; |
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num--; |
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for (j = --i; j < num; j++) |
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S(j) = S(j + 1); |
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} |
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for (i = 0; !i || s->segments[i - 2].x; i += 2) { |
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s->segments[i].y += s->gain_dB; |
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s->segments[i].x *= M_LN10 / 20; |
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s->segments[i].y *= M_LN10 / 20; |
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} |
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#define L(x) s->segments[i - (x)] |
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for (i = 4; s->segments[i - 2].x; i += 2) { |
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double x, y, cx, cy, in1, in2, out1, out2, theta, len, r; |
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L(4).a = 0; |
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L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x); |
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L(2).a = 0; |
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L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x); |
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theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x); |
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len = sqrt(pow(L(2).x - L(4).x, 2.) + pow(L(2).y - L(4).y, 2.)); |
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r = FFMIN(radius, len); |
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L(3).x = L(2).x - r * cos(theta); |
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L(3).y = L(2).y - r * sin(theta); |
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theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x); |
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len = sqrt(pow(L(0).x - L(2).x, 2.) + pow(L(0).y - L(2).y, 2.)); |
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r = FFMIN(radius, len / 2); |
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x = L(2).x + r * cos(theta); |
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y = L(2).y + r * sin(theta); |
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cx = (L(3).x + L(2).x + x) / 3; |
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cy = (L(3).y + L(2).y + y) / 3; |
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L(2).x = x; |
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L(2).y = y; |
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in1 = cx - L(3).x; |
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out1 = cy - L(3).y; |
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in2 = L(2).x - L(3).x; |
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out2 = L(2).y - L(3).y; |
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L(3).a = (out2 / in2 - out1 / in1) / (in2-in1); |
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L(3).b = out1 / in1 - L(3).a * in1; |
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} |
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L(3).x = 0; |
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L(3).y = L(2).y; |
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s->in_min_lin = exp(s->segments[1].x); |
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s->out_min_lin = exp(s->segments[1].y); |
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for (i = 0; i < outlink->channels; i++) { |
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ChanParam *cp = &s->channels[i]; |
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if (cp->attack > 1.0 / sample_rate) |
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cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack)); |
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else |
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cp->attack = 1.0; |
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if (cp->decay > 1.0 / sample_rate) |
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cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay)); |
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else |
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cp->decay = 1.0; |
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cp->volume = pow(10.0, s->initial_volume / 20); |
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} |
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s->delay_samples = s->delay * sample_rate; |
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if (s->delay_samples > 0) { |
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int ret; |
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if ((ret = av_samples_alloc_array_and_samples(&s->delayptrs, NULL, |
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outlink->channels, |
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s->delay_samples, |
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outlink->format, 0)) < 0) |
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return ret; |
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s->compand = compand_delay; |
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outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP; |
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} else { |
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s->compand = compand_nodelay; |
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} |
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return 0; |
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} |
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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CompandContext *s = ctx->priv; |
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return s->compand(ctx, frame); |
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} |
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static int request_frame(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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CompandContext *s = ctx->priv; |
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int ret; |
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ret = ff_request_frame(ctx->inputs[0]); |
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if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count) |
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ret = compand_drain(outlink); |
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return ret; |
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} |
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static const AVFilterPad compand_inputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.filter_frame = filter_frame, |
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}, |
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{ NULL }, |
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}; |
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static const AVFilterPad compand_outputs[] = { |
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{ |
||||
.name = "default", |
||||
.request_frame = request_frame, |
||||
.config_props = config_output, |
||||
.type = AVMEDIA_TYPE_AUDIO, |
||||
}, |
||||
{ NULL }, |
||||
}; |
||||
|
||||
AVFilter avfilter_af_compand = { |
||||
.name = "compand", |
||||
.description = NULL_IF_CONFIG_SMALL("Compress or expand audio dynamic range."), |
||||
.query_formats = query_formats, |
||||
.priv_size = sizeof(CompandContext), |
||||
.priv_class = &compand_class, |
||||
.init = init, |
||||
.uninit = uninit, |
||||
.inputs = compand_inputs, |
||||
.outputs = compand_outputs, |
||||
}; |
Loading…
Reference in new issue