mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
11 years ago
parent
415f1fab8d
commit
d2fc702ace
6 changed files with 439 additions and 2 deletions
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/*
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* Copyright (c) 1998 Juergen Mueller And Sundry Contributors |
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* This source code is freely redistributable and may be used for |
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* any purpose. This copyright notice must be maintained. |
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* Juergen Mueller And Sundry Contributors are not responsible for |
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* the consequences of using this software. |
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* |
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* Copyright (c) 2015 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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* @file |
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* chorus audio filter |
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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 "audio.h" |
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#include "avfilter.h" |
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#include "internal.h" |
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#include "generate_wave_table.h" |
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typedef struct ChorusContext { |
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const AVClass *class; |
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float in_gain, out_gain; |
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char *delays_str; |
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char *decays_str; |
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char *speeds_str; |
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char *depths_str; |
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float *delays; |
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float *decays; |
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float *speeds; |
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float *depths; |
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uint8_t **chorusbuf; |
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int **phase; |
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int *length; |
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int32_t **lookup_table; |
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int *counter; |
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int num_chorus; |
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int max_samples; |
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int channels; |
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int modulation; |
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int fade_out; |
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int64_t next_pts; |
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} ChorusContext; |
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#define OFFSET(x) offsetof(ChorusContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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static const AVOption chorus_options[] = { |
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{ "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A }, |
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{ "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A }, |
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{ "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(chorus); |
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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 fill_items(char *item_str, int *nb_items, float *items) |
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{ |
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char *p, *saveptr = NULL; |
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int i, new_nb_items = 0; |
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p = item_str; |
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for (i = 0; i < *nb_items; 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, "%f", &items[i]) == 1; |
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} |
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*nb_items = new_nb_items; |
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} |
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static av_cold int init(AVFilterContext *ctx) |
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{ |
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ChorusContext *s = ctx->priv; |
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int nb_delays, nb_decays, nb_speeds, nb_depths; |
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if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) { |
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av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n"); |
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return AVERROR(EINVAL); |
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} |
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count_items(s->delays_str, &nb_delays); |
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count_items(s->decays_str, &nb_decays); |
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count_items(s->speeds_str, &nb_speeds); |
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count_items(s->depths_str, &nb_depths); |
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s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays)); |
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s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays)); |
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s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds)); |
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s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths)); |
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if (!s->delays || !s->decays || !s->speeds || !s->depths) |
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return AVERROR(ENOMEM); |
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fill_items(s->delays_str, &nb_delays, s->delays); |
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fill_items(s->decays_str, &nb_decays, s->decays); |
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fill_items(s->speeds_str, &nb_speeds, s->speeds); |
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fill_items(s->depths_str, &nb_depths, s->depths); |
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if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) { |
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av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n"); |
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return AVERROR(EINVAL); |
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} |
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s->num_chorus = nb_delays; |
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if (s->num_chorus < 1) { |
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av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n"); |
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return AVERROR(EINVAL); |
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} |
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s->length = av_calloc(s->num_chorus, sizeof(*s->length)); |
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s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table)); |
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if (!s->length || !s->lookup_table) |
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return AVERROR(ENOMEM); |
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s->next_pts = AV_NOPTS_VALUE; |
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return 0; |
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} |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AVFilterFormats *formats; |
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AVFilterChannelLayouts *layouts; |
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static const enum AVSampleFormat sample_fmts[] = { |
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE |
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}; |
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int ret; |
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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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ret = ff_set_common_channel_layouts(ctx, layouts); |
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if (ret < 0) |
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return ret; |
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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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ret = ff_set_common_formats(ctx, formats); |
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if (ret < 0) |
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return ret; |
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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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return ff_set_common_samplerates(ctx, formats); |
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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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ChorusContext *s = ctx->priv; |
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float sum_in_volume = 1.0; |
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int n; |
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s->channels = outlink->channels; |
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for (n = 0; n < s->num_chorus; n++) { |
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int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0); |
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int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0); |
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s->length[n] = outlink->sample_rate / s->speeds[n]; |
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s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]); |
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if (!s->lookup_table[n]) |
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return AVERROR(ENOMEM); |
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ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n], |
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s->length[n], 0., depth_samples, 0); |
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s->max_samples = FFMAX(s->max_samples, samples); |
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} |
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for (n = 0; n < s->num_chorus; n++) |
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sum_in_volume += s->decays[n]; |
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if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain) |
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av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n"); |
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s->counter = av_calloc(outlink->channels, sizeof(*s->counter)); |
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if (!s->counter) |
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return AVERROR(ENOMEM); |
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s->phase = av_calloc(outlink->channels, sizeof(*s->phase)); |
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if (!s->phase) |
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return AVERROR(ENOMEM); |
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for (n = 0; n < outlink->channels; n++) { |
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s->phase[n] = av_calloc(s->num_chorus, sizeof(int)); |
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if (!s->phase[n]) |
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return AVERROR(ENOMEM); |
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} |
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s->fade_out = s->max_samples; |
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return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL, |
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outlink->channels, |
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s->max_samples, |
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outlink->format, 0); |
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} |
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#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a)) |
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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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ChorusContext *s = ctx->priv; |
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AVFrame *out_frame; |
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int c, i, n; |
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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, frame->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 (c = 0; c < inlink->channels; c++) { |
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const float *src = (const float *)frame->extended_data[c]; |
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float *dst = (float *)out_frame->extended_data[c]; |
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float *chorusbuf = (float *)s->chorusbuf[c]; |
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int *phase = s->phase[c]; |
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for (i = 0; i < frame->nb_samples; i++) { |
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float out, in = src[i]; |
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out = in * s->in_gain; |
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for (n = 0; n < s->num_chorus; n++) { |
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out += chorusbuf[MOD(s->max_samples + s->counter[c] - |
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s->lookup_table[n][phase[n]], |
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s->max_samples)] * s->decays[n]; |
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phase[n] = MOD(phase[n] + 1, s->length[n]); |
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} |
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out *= s->out_gain; |
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dst[i] = out; |
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chorusbuf[s->counter[c]] = in; |
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s->counter[c] = MOD(s->counter[c] + 1, s->max_samples); |
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} |
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} |
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s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base); |
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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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static int request_frame(AVFilterLink *outlink) |
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{ |
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AVFilterContext *ctx = outlink->src; |
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ChorusContext *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->fade_out) { |
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int nb_samples = FFMIN(s->fade_out, 2048); |
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AVFrame *frame; |
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frame = ff_get_audio_buffer(outlink, nb_samples); |
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if (!frame) |
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return AVERROR(ENOMEM); |
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s->fade_out -= nb_samples; |
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av_samples_set_silence(frame->extended_data, 0, |
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frame->nb_samples, |
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outlink->channels, |
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frame->format); |
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frame->pts = s->next_pts; |
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if (s->next_pts != AV_NOPTS_VALUE) |
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
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ret = filter_frame(ctx->inputs[0], frame); |
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} |
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return ret; |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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ChorusContext *s = ctx->priv; |
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int n; |
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av_freep(&s->delays); |
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av_freep(&s->decays); |
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av_freep(&s->speeds); |
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av_freep(&s->depths); |
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if (s->chorusbuf) |
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av_freep(&s->chorusbuf[0]); |
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av_freep(&s->chorusbuf); |
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if (s->phase) |
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for (n = 0; n < s->channels; n++) |
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av_freep(&s->phase[n]); |
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av_freep(&s->phase); |
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av_freep(&s->counter); |
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av_freep(&s->length); |
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if (s->lookup_table) |
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for (n = 0; n < s->num_chorus; n++) |
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av_freep(&s->lookup_table[n]); |
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av_freep(&s->lookup_table); |
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} |
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static const AVFilterPad chorus_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 chorus_outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.request_frame = request_frame, |
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.config_props = config_output, |
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}, |
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{ NULL } |
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}; |
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AVFilter ff_af_chorus = { |
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.name = "chorus", |
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.description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."), |
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.query_formats = query_formats, |
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.priv_size = sizeof(ChorusContext), |
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.priv_class = &chorus_class, |
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.init = init, |
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.uninit = uninit, |
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.inputs = chorus_inputs, |
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.outputs = chorus_outputs, |
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}; |
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